initial commit

.gitignore

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+.ipynb_checkpoints

app.py

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+# ruff: noqa: E402
+# Above allows ruff to ignore E402: module level import not at top of file
+import os
+os.system('git clone https://github.com/NVIDIA/BigVGAN.git')
+
+
+import re
+import tempfile
+from collections import OrderedDict
+from importlib.resources import files
+
+import click
+import gradio as gr
+import numpy as np
+import soundfile as sf
+import torchaudio
+from cached_path import cached_path
+from transformers import AutoModelForCausalLM, AutoTokenizer
+
+try:
+    import spaces
+
+    USING_SPACES = True
+except ImportError:
+    USING_SPACES = False
+
+
+def gpu_decorator(func):
+    if USING_SPACES:
+        return spaces.GPU(func)
+    else:
+        return func
+
+
+from model import DiT, UNetT
+from infer_utils import (
+    load_vocoder,
+    load_model,
+    preprocess_ref_audio_text,
+    infer_process,
+    remove_silence_for_generated_wav,
+    save_spectrogram,
+)
+
+
+DEFAULT_TTS_MODEL = "F5-TTS"
+tts_model_choice = DEFAULT_TTS_MODEL
+
+
+# load models
+
+from huggingface_hub import hf_hub_download
+import joblib
+
+model_file = joblib.load(
+    hf_hub_download(repo_id="attashe/F5-TTS-Ru-finetune", filename="model_last_bigvgan.safetensors")
+)
+
+vocab_file = joblib.load(
+    hf_hub_download(repo_id="attashe/F5-TTS-Ru-finetune", filename="vocab.txt")
+)
+print(f"Using model file: {model_file} and vocab: {vocab_file}")
+
+vocoder = load_vocoder(vocoder_name="bigvgan")
+
+
+def load_f5tts(ckpt_path=str(cached_path("hf://SWivid/F5-TTS/F5TTS_Base/model_1200000.safetensors"))):
+    ckpt_path = model_file
+    vocab_path = vocab_file
+    F5TTS_model_cfg = dict(dim=1024, depth=22, heads=16, ff_mult=2, text_dim=512, conv_layers=4)
+    return load_model(DiT, F5TTS_model_cfg, ckpt_path, vocab_file=vocab_path)
+
+
+def load_e2tts(ckpt_path=str(cached_path("hf://SWivid/E2-TTS/E2TTS_Base/model_1200000.safetensors"))):
+    E2TTS_model_cfg = dict(dim=1024, depth=24, heads=16, ff_mult=4)
+    return load_model(UNetT, E2TTS_model_cfg, ckpt_path)
+
+
+def load_custom(ckpt_path: str, vocab_path="", model_cfg=None):
+    ckpt_path, vocab_path = ckpt_path.strip(), vocab_path.strip()
+    if ckpt_path.startswith("hf://"):
+        ckpt_path = str(cached_path(ckpt_path))
+    if vocab_path.startswith("hf://"):
+        vocab_path = str(cached_path(vocab_path))
+    if model_cfg is None:
+        model_cfg = dict(dim=1024, depth=22, heads=16, ff_mult=2, text_dim=512, conv_layers=4)
+    return load_model(DiT, model_cfg, ckpt_path, vocab_file=vocab_path)
+
+
+F5TTS_ema_model = load_f5tts()
+E2TTS_ema_model = load_e2tts() if USING_SPACES else None
+custom_ema_model, pre_custom_path = None, ""
+
+chat_model_state = None
+chat_tokenizer_state = None
+
+
+@gpu_decorator
+def generate_response(messages, model, tokenizer):
+    """Generate response using Qwen"""
+    text = tokenizer.apply_chat_template(
+        messages,
+        tokenize=False,
+        add_generation_prompt=True,
+    )
+
+    model_inputs = tokenizer([text], return_tensors="pt").to(model.device)
+    generated_ids = model.generate(
+        **model_inputs,
+        max_new_tokens=512,
+        temperature=0.7,
+        top_p=0.95,
+    )
+
+    generated_ids = [
+        output_ids[len(input_ids) :] for input_ids, output_ids in zip(model_inputs.input_ids, generated_ids)
+    ]
+    return tokenizer.batch_decode(generated_ids, skip_special_tokens=True)[0]
+
+
+@gpu_decorator
+def infer(
+    ref_audio_orig, ref_text, gen_text, model, remove_silence, cross_fade_duration=0.15, speed=1, show_info=gr.Info
+):
+    ref_audio, ref_text = preprocess_ref_audio_text(ref_audio_orig, ref_text, show_info=show_info)
+    gen_text = gen_text.lower().strip()
+    ref_text = ref_text.lower().strip()
+
+    if model == "F5-TTS":
+        ema_model = F5TTS_ema_model
+    elif model == "E2-TTS":
+        global E2TTS_ema_model
+        if E2TTS_ema_model is None:
+            show_info("Loading E2-TTS model...")
+            E2TTS_ema_model = load_e2tts()
+        ema_model = E2TTS_ema_model
+    elif isinstance(model, list) and model[0] == "Custom":
+        assert not USING_SPACES, "Only official checkpoints allowed in Spaces."
+        global custom_ema_model, pre_custom_path
+        if pre_custom_path != model[1]:
+            show_info("Loading Custom TTS model...")
+            custom_ema_model = load_custom(model[1], vocab_path=model[2])
+            pre_custom_path = model[1]
+        ema_model = custom_ema_model
+
+    final_wave, final_sample_rate, combined_spectrogram = infer_process(
+        ref_audio,
+        ref_text,
+        gen_text,
+        ema_model,
+        vocoder,
+        cross_fade_duration=cross_fade_duration,
+        speed=speed,
+        show_info=show_info,
+        progress=gr.Progress(),
+    )
+
+    # Remove silence
+    if remove_silence:
+        with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as f:
+            sf.write(f.name, final_wave, final_sample_rate)
+            remove_silence_for_generated_wav(f.name)
+            final_wave, _ = torchaudio.load(f.name)
+        final_wave = final_wave.squeeze().cpu().numpy()
+
+    # Save the spectrogram
+    with tempfile.NamedTemporaryFile(suffix=".png", delete=False) as tmp_spectrogram:
+        spectrogram_path = tmp_spectrogram.name
+        save_spectrogram(combined_spectrogram, spectrogram_path)
+
+    return (final_sample_rate, final_wave), spectrogram_path, ref_text
+
+
+with gr.Blocks() as app_credits:
+    gr.Markdown("""
+# Credits
+
+* [mrfakename](https://github.com/fakerybakery) for the original [online demo](https://huggingface.co/spaces/mrfakename/E2-F5-TTS)
+* [RootingInLoad](https://github.com/RootingInLoad) for initial chunk generation and podcast app exploration
+* [jpgallegoar](https://github.com/jpgallegoar) for multiple speech-type generation & voice chat
+""")
+with gr.Blocks() as app_tts:
+    gr.Markdown("# Batched TTS")
+    ref_audio_input = gr.Audio(label="Reference Audio", type="filepath")
+    gen_text_input = gr.Textbox(label="Text to Generate", lines=10)
+    generate_btn = gr.Button("Synthesize", variant="primary")
+    with gr.Accordion("Advanced Settings", open=False):
+        ref_text_input = gr.Textbox(
+            label="Reference Text",
+            info="Leave blank to automatically transcribe the reference audio. If you enter text it will override automatic transcription.",
+            lines=2,
+        )
+        remove_silence = gr.Checkbox(
+            label="Remove Silences",
+            info="The model tends to produce silences, especially on longer audio. We can manually remove silences if needed. Note that this is an experimental feature and may produce strange results. This will also increase generation time.",
+            value=False,
+        )
+        speed_slider = gr.Slider(
+            label="Speed",
+            minimum=0.3,
+            maximum=2.0,
+            value=1.0,
+            step=0.1,
+            info="Adjust the speed of the audio.",
+        )
+        cross_fade_duration_slider = gr.Slider(
+            label="Cross-Fade Duration (s)",
+            minimum=0.0,
+            maximum=1.0,
+            value=0.15,
+            step=0.01,
+            info="Set the duration of the cross-fade between audio clips.",
+        )
+
+    audio_output = gr.Audio(label="Synthesized Audio")
+    spectrogram_output = gr.Image(label="Spectrogram")
+
+    @gpu_decorator
+    def basic_tts(
+        ref_audio_input,
+        ref_text_input,
+        gen_text_input,
+        remove_silence,
+        cross_fade_duration_slider,
+        speed_slider,
+    ):
+        audio_out, spectrogram_path, ref_text_out = infer(
+            ref_audio_input,
+            ref_text_input,
+            gen_text_input,
+            tts_model_choice,
+            remove_silence,
+            cross_fade_duration_slider,
+            speed_slider,
+        )
+        return audio_out, spectrogram_path, gr.update(value=ref_text_out)
+
+    generate_btn.click(
+        basic_tts,
+        inputs=[
+            ref_audio_input,
+            ref_text_input,
+            gen_text_input,
+            remove_silence,
+            cross_fade_duration_slider,
+            speed_slider,
+        ],
+        outputs=[audio_output, spectrogram_output, ref_text_input],
+    )
+
+
+def parse_speechtypes_text(gen_text):
+    # Pattern to find {speechtype}
+    pattern = r"\{(.*?)\}"
+
+    # Split the text by the pattern
+    tokens = re.split(pattern, gen_text)
+
+    segments = []
+
+    current_style = "Regular"
+
+    for i in range(len(tokens)):
+        if i % 2 == 0:
+            # This is text
+            text = tokens[i].strip()
+            if text:
+                segments.append({"style": current_style, "text": text})
+        else:
+            # This is style
+            style = tokens[i].strip()
+            current_style = style
+
+    return segments
+
+
+with gr.Blocks() as app_multistyle:
+    # New section for multistyle generation
+    gr.Markdown(
+        """
+    # Multiple Speech-Type Generation
+
+    This section allows you to generate multiple speech types or multiple people's voices. Enter your text in the format shown below, and the system will generate speech using the appropriate type. If unspecified, the model will use the regular speech type. The current speech type will be used until the next speech type is specified.
+    """
+    )
+
+    with gr.Row():
+        gr.Markdown(
+            """
+            **Example Input:**                                                                      
+            {Regular} Hello, I'd like to order a sandwich please.                                                         
+            {Surprised} What do you mean you're out of bread?                                                                      
+            {Sad} I really wanted a sandwich though...                                                              
+            {Angry} You know what, darn you and your little shop!                                                                       
+            {Whisper} I'll just go back home and cry now.                                                                           
+            {Shouting} Why me?!                                                                         
+            """
+        )
+
+        gr.Markdown(
+            """
+            **Example Input 2:**                                                                                
+            {Speaker1_Happy} Hello, I'd like to order a sandwich please.                                                            
+            {Speaker2_Regular} Sorry, we're out of bread.                                                                                
+            {Speaker1_Sad} I really wanted a sandwich though...                                                                             
+            {Speaker2_Whisper} I'll give you the last one I was hiding.                                                                     
+            """
+        )
+
+    gr.Markdown(
+        "Upload different audio clips for each speech type. The first speech type is mandatory. You can add additional speech types by clicking the 'Add Speech Type' button."
+    )
+
+    # Regular speech type (mandatory)
+    with gr.Row():
+        with gr.Column():
+            regular_name = gr.Textbox(value="Regular", label="Speech Type Name")
+            regular_insert = gr.Button("Insert Label", variant="secondary")
+        regular_audio = gr.Audio(label="Regular Reference Audio", type="filepath")
+        regular_ref_text = gr.Textbox(label="Reference Text (Regular)", lines=2)
+
+    # Regular speech type (max 100)
+    max_speech_types = 100
+    speech_type_rows = []  # 99
+    speech_type_names = [regular_name]  # 100
+    speech_type_audios = [regular_audio]  # 100
+    speech_type_ref_texts = [regular_ref_text]  # 100
+    speech_type_delete_btns = []  # 99
+    speech_type_insert_btns = [regular_insert]  # 100
+
+    # Additional speech types (99 more)
+    for i in range(max_speech_types - 1):
+        with gr.Row(visible=False) as row:
+            with gr.Column():
+                name_input = gr.Textbox(label="Speech Type Name")
+                delete_btn = gr.Button("Delete Type", variant="secondary")
+                insert_btn = gr.Button("Insert Label", variant="secondary")
+            audio_input = gr.Audio(label="Reference Audio", type="filepath")
+            ref_text_input = gr.Textbox(label="Reference Text", lines=2)
+        speech_type_rows.append(row)
+        speech_type_names.append(name_input)
+        speech_type_audios.append(audio_input)
+        speech_type_ref_texts.append(ref_text_input)
+        speech_type_delete_btns.append(delete_btn)
+        speech_type_insert_btns.append(insert_btn)
+
+    # Button to add speech type
+    add_speech_type_btn = gr.Button("Add Speech Type")
+
+    # Keep track of current number of speech types
+    speech_type_count = gr.State(value=1)
+
+    # Function to add a speech type
+    def add_speech_type_fn(speech_type_count):
+        if speech_type_count < max_speech_types:
+            speech_type_count += 1
+            # Prepare updates for the rows
+            row_updates = []
+            for i in range(1, max_speech_types):
+                if i < speech_type_count:
+                    row_updates.append(gr.update(visible=True))
+                else:
+                    row_updates.append(gr.update())
+        else:
+            # Optionally, show a warning
+            row_updates = [gr.update() for _ in range(1, max_speech_types)]
+        return [speech_type_count] + row_updates
+
+    add_speech_type_btn.click(
+        add_speech_type_fn, inputs=speech_type_count, outputs=[speech_type_count] + speech_type_rows
+    )
+
+    # Function to delete a speech type
+    def make_delete_speech_type_fn(index):
+        def delete_speech_type_fn(speech_type_count):
+            # Prepare updates
+            row_updates = []
+
+            for i in range(1, max_speech_types):
+                if i == index:
+                    row_updates.append(gr.update(visible=False))
+                else:
+                    row_updates.append(gr.update())
+
+            speech_type_count = max(1, speech_type_count)
+
+            return [speech_type_count] + row_updates
+
+        return delete_speech_type_fn
+
+    # Update delete button clicks
+    for i, delete_btn in enumerate(speech_type_delete_btns):
+        delete_fn = make_delete_speech_type_fn(i)
+        delete_btn.click(delete_fn, inputs=speech_type_count, outputs=[speech_type_count] + speech_type_rows)
+
+    # Text input for the prompt
+    gen_text_input_multistyle = gr.Textbox(
+        label="Text to Generate",
+        lines=10,
+        placeholder="Enter the script with speaker names (or emotion types) at the start of each block, e.g.:\n\n{Regular} Hello, I'd like to order a sandwich please.\n{Surprised} What do you mean you're out of bread?\n{Sad} I really wanted a sandwich though...\n{Angry} You know what, darn you and your little shop!\n{Whisper} I'll just go back home and cry now.\n{Shouting} Why me?!",
+    )
+
+    def make_insert_speech_type_fn(index):
+        def insert_speech_type_fn(current_text, speech_type_name):
+            current_text = current_text or ""
+            speech_type_name = speech_type_name or "None"
+            updated_text = current_text + f"{{{speech_type_name}}} "
+            return gr.update(value=updated_text)
+
+        return insert_speech_type_fn
+
+    for i, insert_btn in enumerate(speech_type_insert_btns):
+        insert_fn = make_insert_speech_type_fn(i)
+        insert_btn.click(
+            insert_fn,
+            inputs=[gen_text_input_multistyle, speech_type_names[i]],
+            outputs=gen_text_input_multistyle,
+        )
+
+    with gr.Accordion("Advanced Settings", open=False):
+        remove_silence_multistyle = gr.Checkbox(
+            label="Remove Silences",
+            value=True,
+        )
+
+    # Generate button
+    generate_multistyle_btn = gr.Button("Generate Multi-Style Speech", variant="primary")
+
+    # Output audio
+    audio_output_multistyle = gr.Audio(label="Synthesized Audio")
+
+    @gpu_decorator
+    def generate_multistyle_speech(
+        gen_text,
+        *args,
+    ):
+        speech_type_names_list = args[:max_speech_types]
+        speech_type_audios_list = args[max_speech_types : 2 * max_speech_types]
+        speech_type_ref_texts_list = args[2 * max_speech_types : 3 * max_speech_types]
+        remove_silence = args[3 * max_speech_types]
+        # Collect the speech types and their audios into a dict
+        speech_types = OrderedDict()
+
+        ref_text_idx = 0
+        for name_input, audio_input, ref_text_input in zip(
+            speech_type_names_list, speech_type_audios_list, speech_type_ref_texts_list
+        ):
+            if name_input and audio_input:
+                speech_types[name_input] = {"audio": audio_input, "ref_text": ref_text_input}
+            else:
+                speech_types[f"@{ref_text_idx}@"] = {"audio": "", "ref_text": ""}
+            ref_text_idx += 1
+
+        # Parse the gen_text into segments
+        segments = parse_speechtypes_text(gen_text)
+
+        # For each segment, generate speech
+        generated_audio_segments = []
+        current_style = "Regular"
+
+        for segment in segments:
+            style = segment["style"]
+            text = segment["text"]
+
+            if style in speech_types:
+                current_style = style
+            else:
+                # If style not available, default to Regular
+                current_style = "Regular"
+
+            ref_audio = speech_types[current_style]["audio"]
+            ref_text = speech_types[current_style].get("ref_text", "")
+
+            # Generate speech for this segment
+            audio_out, _, ref_text_out = infer(
+                ref_audio, ref_text, text, tts_model_choice, remove_silence, 0, show_info=print
+            )  # show_info=print no pull to top when generating
+            sr, audio_data = audio_out
+
+            generated_audio_segments.append(audio_data)
+            speech_types[current_style]["ref_text"] = ref_text_out
+
+        # Concatenate all audio segments
+        if generated_audio_segments:
+            final_audio_data = np.concatenate(generated_audio_segments)
+            return [(sr, final_audio_data)] + [
+                gr.update(value=speech_types[style]["ref_text"]) for style in speech_types
+            ]
+        else:
+            gr.Warning("No audio generated.")
+            return [None] + [gr.update(value=speech_types[style]["ref_text"]) for style in speech_types]
+
+    generate_multistyle_btn.click(
+        generate_multistyle_speech,
+        inputs=[
+            gen_text_input_multistyle,
+        ]
+        + speech_type_names
+        + speech_type_audios
+        + speech_type_ref_texts
+        + [
+            remove_silence_multistyle,
+        ],
+        outputs=[audio_output_multistyle] + speech_type_ref_texts,
+    )
+
+    # Validation function to disable Generate button if speech types are missing
+    def validate_speech_types(gen_text, regular_name, *args):
+        speech_type_names_list = args[:max_speech_types]
+
+        # Collect the speech types names
+        speech_types_available = set()
+        if regular_name:
+            speech_types_available.add(regular_name)
+        for name_input in speech_type_names_list:
+            if name_input:
+                speech_types_available.add(name_input)
+
+        # Parse the gen_text to get the speech types used
+        segments = parse_speechtypes_text(gen_text)
+        speech_types_in_text = set(segment["style"] for segment in segments)
+
+        # Check if all speech types in text are available
+        missing_speech_types = speech_types_in_text - speech_types_available
+
+        if missing_speech_types:
+            # Disable the generate button
+            return gr.update(interactive=False)
+        else:
+            # Enable the generate button
+            return gr.update(interactive=True)
+
+    gen_text_input_multistyle.change(
+        validate_speech_types,
+        inputs=[gen_text_input_multistyle, regular_name] + speech_type_names,
+        outputs=generate_multistyle_btn,
+    )
+
+
+with gr.Blocks() as app_chat:
+    gr.Markdown(
+        """
+# Voice Chat
+Have a conversation with an AI using your reference voice! 
+1. Upload a reference audio clip and optionally its transcript.
+2. Load the chat model.
+3. Record your message through your microphone.
+4. The AI will respond using the reference voice.
+"""
+    )
+
+    if not USING_SPACES:
+        load_chat_model_btn = gr.Button("Load Chat Model", variant="primary")
+
+        chat_interface_container = gr.Column(visible=False)
+
+        @gpu_decorator
+        def load_chat_model():
+            global chat_model_state, chat_tokenizer_state
+            if chat_model_state is None:
+                show_info = gr.Info
+                show_info("Loading chat model...")
+                model_name = "Qwen/Qwen2.5-3B-Instruct"
+                chat_model_state = AutoModelForCausalLM.from_pretrained(
+                    model_name, torch_dtype="auto", device_map="auto"
+                )
+                chat_tokenizer_state = AutoTokenizer.from_pretrained(model_name)
+                show_info("Chat model loaded.")
+
+            return gr.update(visible=False), gr.update(visible=True)
+
+        load_chat_model_btn.click(load_chat_model, outputs=[load_chat_model_btn, chat_interface_container])
+
+    else:
+        chat_interface_container = gr.Column()
+
+        if chat_model_state is None:
+            model_name = "Qwen/Qwen2.5-3B-Instruct"
+            chat_model_state = AutoModelForCausalLM.from_pretrained(model_name, torch_dtype="auto", device_map="auto")
+            chat_tokenizer_state = AutoTokenizer.from_pretrained(model_name)
+
+    with chat_interface_container:
+        with gr.Row():
+            with gr.Column():
+                ref_audio_chat = gr.Audio(label="Reference Audio", type="filepath")
+            with gr.Column():
+                with gr.Accordion("Advanced Settings", open=False):
+                    remove_silence_chat = gr.Checkbox(
+                        label="Remove Silences",
+                        value=True,
+                    )
+                    ref_text_chat = gr.Textbox(
+                        label="Reference Text",
+                        info="Optional: Leave blank to auto-transcribe",
+                        lines=2,
+                    )
+                    system_prompt_chat = gr.Textbox(
+                        label="System Prompt",
+                        value="You are not an AI assistant, you are whoever the user says you are. You must stay in character. Keep your responses concise since they will be spoken out loud.",
+                        lines=2,
+                    )
+
+        chatbot_interface = gr.Chatbot(label="Conversation")
+
+        with gr.Row():
+            with gr.Column():
+                audio_input_chat = gr.Microphone(
+                    label="Speak your message",
+                    type="filepath",
+                )
+                audio_output_chat = gr.Audio(autoplay=True)
+            with gr.Column():
+                text_input_chat = gr.Textbox(
+                    label="Type your message",
+                    lines=1,
+                )
+                send_btn_chat = gr.Button("Send Message")
+                clear_btn_chat = gr.Button("Clear Conversation")
+
+        conversation_state = gr.State(
+            value=[
+                {
+                    "role": "system",
+                    "content": "You are not an AI assistant, you are whoever the user says you are. You must stay in character. Keep your responses concise since they will be spoken out loud.",
+                }
+            ]
+        )
+
+        # Modify process_audio_input to use model and tokenizer from state
+        @gpu_decorator
+        def process_audio_input(audio_path, text, history, conv_state):
+            """Handle audio or text input from user"""
+
+            if not audio_path and not text.strip():
+                return history, conv_state, ""
+
+            if audio_path:
+                text = preprocess_ref_audio_text(audio_path, text)[1]
+
+            if not text.strip():
+                return history, conv_state, ""
+
+            conv_state.append({"role": "user", "content": text})
+            history.append((text, None))
+
+            response = generate_response(conv_state, chat_model_state, chat_tokenizer_state)
+
+            conv_state.append({"role": "assistant", "content": response})
+            history[-1] = (text, response)
+
+            return history, conv_state, ""
+
+        @gpu_decorator
+        def generate_audio_response(history, ref_audio, ref_text, remove_silence):
+            """Generate TTS audio for AI response"""
+            if not history or not ref_audio:
+                return None
+
+            last_user_message, last_ai_response = history[-1]
+            if not last_ai_response:
+                return None
+
+            audio_result, _, ref_text_out = infer(
+                ref_audio,
+                ref_text,
+                last_ai_response,
+                tts_model_choice,
+                remove_silence,
+                cross_fade_duration=0.15,
+                speed=1.0,
+                show_info=print,  # show_info=print no pull to top when generating
+            )
+            return audio_result, gr.update(value=ref_text_out)
+
+        def clear_conversation():
+            """Reset the conversation"""
+            return [], [
+                {
+                    "role": "system",
+                    "content": "You are not an AI assistant, you are whoever the user says you are. You must stay in character. Keep your responses concise since they will be spoken out loud.",
+                }
+            ]
+
+        def update_system_prompt(new_prompt):
+            """Update the system prompt and reset the conversation"""
+            new_conv_state = [{"role": "system", "content": new_prompt}]
+            return [], new_conv_state
+
+        # Handle audio input
+        audio_input_chat.stop_recording(
+            process_audio_input,
+            inputs=[audio_input_chat, text_input_chat, chatbot_interface, conversation_state],
+            outputs=[chatbot_interface, conversation_state],
+        ).then(
+            generate_audio_response,
+            inputs=[chatbot_interface, ref_audio_chat, ref_text_chat, remove_silence_chat],
+            outputs=[audio_output_chat, ref_text_chat],
+        ).then(
+            lambda: None,
+            None,
+            audio_input_chat,
+        )
+
+        # Handle text input
+        text_input_chat.submit(
+            process_audio_input,
+            inputs=[audio_input_chat, text_input_chat, chatbot_interface, conversation_state],
+            outputs=[chatbot_interface, conversation_state],
+        ).then(
+            generate_audio_response,
+            inputs=[chatbot_interface, ref_audio_chat, ref_text_chat, remove_silence_chat],
+            outputs=[audio_output_chat, ref_text_chat],
+        ).then(
+            lambda: None,
+            None,
+            text_input_chat,
+        )
+
+        # Handle send button
+        send_btn_chat.click(
+            process_audio_input,
+            inputs=[audio_input_chat, text_input_chat, chatbot_interface, conversation_state],
+            outputs=[chatbot_interface, conversation_state],
+        ).then(
+            generate_audio_response,
+            inputs=[chatbot_interface, ref_audio_chat, ref_text_chat, remove_silence_chat],
+            outputs=[audio_output_chat, ref_text_chat],
+        ).then(
+            lambda: None,
+            None,
+            text_input_chat,
+        )
+
+        # Handle clear button
+        clear_btn_chat.click(
+            clear_conversation,
+            outputs=[chatbot_interface, conversation_state],
+        )
+
+        # Handle system prompt change and reset conversation
+        system_prompt_chat.change(
+            update_system_prompt,
+            inputs=system_prompt_chat,
+            outputs=[chatbot_interface, conversation_state],
+        )
+
+
+with gr.Blocks() as app:
+    gr.Markdown(
+        """
+# E2/F5 TTS
+
+This is a local web UI for F5 TTS with advanced batch processing support. This app supports the following TTS models:
+
+* [F5-TTS](https://arxiv.org/abs/2410.06885) (A Fairytaler that Fakes Fluent and Faithful Speech with Flow Matching)
+* [E2 TTS](https://arxiv.org/abs/2406.18009) (Embarrassingly Easy Fully Non-Autoregressive Zero-Shot TTS)
+
+The checkpoints currently support English and Chinese.
+
+If you're having issues, try converting your reference audio to WAV or MP3, clipping it to 15s with  ✂  in the bottom right corner (otherwise might have non-optimal auto-trimmed result).
+
+**NOTE: Reference text will be automatically transcribed with Whisper if not provided. For best results, keep your reference clips short (<15s). Ensure the audio is fully uploaded before generating.**
+"""
+    )
+
+    last_used_custom = files("f5_tts").joinpath("infer/.cache/last_used_custom.txt")
+
+    def load_last_used_custom():
+        try:
+            with open(last_used_custom, "r") as f:
+                return f.read().split(",")
+        except FileNotFoundError:
+            last_used_custom.parent.mkdir(parents=True, exist_ok=True)
+            return [
+                "hf://SWivid/F5-TTS/F5TTS_Base/model_1200000.safetensors",
+                "hf://SWivid/F5-TTS/F5TTS_Base/vocab.txt",
+            ]
+
+    def switch_tts_model(new_choice):
+        global tts_model_choice
+        if new_choice == "Custom":  # override in case webpage is refreshed
+            custom_ckpt_path, custom_vocab_path = load_last_used_custom()
+            tts_model_choice = ["Custom", custom_ckpt_path, custom_vocab_path]
+            return gr.update(visible=True, value=custom_ckpt_path), gr.update(visible=True, value=custom_vocab_path)
+        else:
+            tts_model_choice = new_choice
+            return gr.update(visible=False), gr.update(visible=False)
+
+    def set_custom_model(custom_ckpt_path, custom_vocab_path):
+        global tts_model_choice
+        tts_model_choice = ["Custom", custom_ckpt_path, custom_vocab_path]
+        with open(last_used_custom, "w") as f:
+            f.write(f"{custom_ckpt_path},{custom_vocab_path}")
+
+    with gr.Row():
+        if not USING_SPACES:
+            choose_tts_model = gr.Radio(
+                choices=[DEFAULT_TTS_MODEL, "E2-TTS", "Custom"], label="Choose TTS Model", value=DEFAULT_TTS_MODEL
+            )
+        else:
+            choose_tts_model = gr.Radio(
+                choices=[DEFAULT_TTS_MODEL, "E2-TTS"], label="Choose TTS Model", value=DEFAULT_TTS_MODEL
+            )
+        custom_ckpt_path = gr.Dropdown(
+            choices=["hf://SWivid/F5-TTS/F5TTS_Base/model_1200000.safetensors"],
+            value=load_last_used_custom()[0],
+            allow_custom_value=True,
+            label="MODEL CKPT: local_path | hf://user_id/repo_id/model_ckpt",
+            visible=False,
+        )
+        custom_vocab_path = gr.Dropdown(
+            choices=["hf://SWivid/F5-TTS/F5TTS_Base/vocab.txt"],
+            value=load_last_used_custom()[1],
+            allow_custom_value=True,
+            label="VOCAB FILE: local_path | hf://user_id/repo_id/vocab_file",
+            visible=False,
+        )
+
+    choose_tts_model.change(
+        switch_tts_model,
+        inputs=[choose_tts_model],
+        outputs=[custom_ckpt_path, custom_vocab_path],
+        show_progress="hidden",
+    )
+    custom_ckpt_path.change(
+        set_custom_model,
+        inputs=[custom_ckpt_path, custom_vocab_path],
+        show_progress="hidden",
+    )
+    custom_vocab_path.change(
+        set_custom_model,
+        inputs=[custom_ckpt_path, custom_vocab_path],
+        show_progress="hidden",
+    )
+
+    gr.TabbedInterface(
+        [app_tts, app_multistyle, app_chat, app_credits],
+        ["Basic-TTS", "Multi-Speech", "Voice-Chat", "Credits"],
+    )
+
+
+@click.command()
+@click.option("--port", "-p", default=None, type=int, help="Port to run the app on")
+@click.option("--host", "-H", default=None, help="Host to run the app on")
+@click.option(
+    "--share",
+    "-s",
+    default=False,
+    is_flag=True,
+    help="Share the app via Gradio share link",
+)
+@click.option("--api", "-a", default=True, is_flag=True, help="Allow API access")
+@click.option(
+    "--root_path",
+    "-r",
+    default=None,
+    type=str,
+    help='The root path (or "mount point") of the application, if it\'s not served from the root ("/") of the domain. Often used when the application is behind a reverse proxy that forwards requests to the application, e.g. set "/myapp" or full URL for application served at "https://example.com/myapp".',
+)
+def main(port, host, share, api, root_path):
+    global app
+    print("Starting app...")
+    app.queue(api_open=api).launch(server_name=host, server_port=port, share=share, show_api=api, root_path=root_path)
+
+
+if __name__ == "__main__":
+    if not USING_SPACES:
+        main()
+    else:
+        app.queue().launch()

infer_utils.py

@@ -0,0 +1,543 @@
+# A unified script for inference process
+# Make adjustments inside functions, and consider both gradio and cli scripts if need to change func output format
+import os
+import sys
+
+os.environ["PYTOCH_ENABLE_MPS_FALLBACK"] = "1"  # for MPS device compatibility
+sys.path.append(f"../../{os.path.dirname(os.path.abspath(__file__))}/third_party/BigVGAN/")
+
+import hashlib
+import re
+import tempfile
+from importlib.resources import files
+
+import matplotlib
+
+matplotlib.use("Agg")
+
+import matplotlib.pylab as plt
+import numpy as np
+import torch
+import torchaudio
+import tqdm
+from huggingface_hub import snapshot_download, hf_hub_download
+from pydub import AudioSegment, silence
+from transformers import pipeline
+from vocos import Vocos
+
+from model import CFM
+from model_utils import (
+    get_tokenizer,
+    convert_char_to_pinyin,
+)
+
+_ref_audio_cache = {}
+
+device = "cuda" if torch.cuda.is_available() else "mps" if torch.backends.mps.is_available() else "cpu"
+
+# -----------------------------------------
+
+target_sample_rate = 24000
+n_mel_channels = 100
+hop_length = 256
+win_length = 1024
+n_fft = 1024
+mel_spec_type = "bigvgan"
+target_rms = 0.1
+cross_fade_duration = 0.15
+ode_method = "euler"
+nfe_step = 32  # 16, 32
+cfg_strength = 2.0
+sway_sampling_coef = -1.0
+speed = 1.0
+fix_duration = None
+
+# -----------------------------------------
+
+
+# chunk text into smaller pieces
+
+
+def chunk_text(text, max_chars=135):
+    """
+    Splits the input text into chunks, each with a maximum number of characters.
+
+    Args:
+        text (str): The text to be split.
+        max_chars (int): The maximum number of characters per chunk.
+
+    Returns:
+        List[str]: A list of text chunks.
+    """
+    chunks = []
+    current_chunk = ""
+    # Split the text into sentences based on punctuation followed by whitespace
+    sentences = re.split(r"(?<=[;:,.!?])\s+|(?<=[；：，。！？])", text)
+
+    for sentence in sentences:
+        if len(current_chunk.encode("utf-8")) + len(sentence.encode("utf-8")) <= max_chars:
+            current_chunk += sentence + " " if sentence and len(sentence[-1].encode("utf-8")) == 1 else sentence
+        else:
+            if current_chunk:
+                chunks.append(current_chunk.strip())
+            current_chunk = sentence + " " if sentence and len(sentence[-1].encode("utf-8")) == 1 else sentence
+
+    if current_chunk:
+        chunks.append(current_chunk.strip())
+
+    return chunks
+
+
+# load vocoder
+def load_vocoder(vocoder_name="vocos", is_local=False, local_path="", device=device, hf_cache_dir=None):
+    if vocoder_name == "vocos":
+        # vocoder = Vocos.from_pretrained("charactr/vocos-mel-24khz").to(device)
+        if is_local:
+            print(f"Load vocos from local path {local_path}")
+            config_path = f"{local_path}/config.yaml"
+            model_path = f"{local_path}/pytorch_model.bin"
+        else:
+            print("Download Vocos from huggingface charactr/vocos-mel-24khz")
+            repo_id = "charactr/vocos-mel-24khz"
+            config_path = hf_hub_download(repo_id=repo_id, cache_dir=hf_cache_dir, filename="config.yaml")
+            model_path = hf_hub_download(repo_id=repo_id, cache_dir=hf_cache_dir, filename="pytorch_model.bin")
+        vocoder = Vocos.from_hparams(config_path)
+        state_dict = torch.load(model_path, map_location="cpu", weights_only=True)
+        from vocos.feature_extractors import EncodecFeatures
+
+        if isinstance(vocoder.feature_extractor, EncodecFeatures):
+            encodec_parameters = {
+                "feature_extractor.encodec." + key: value
+                for key, value in vocoder.feature_extractor.encodec.state_dict().items()
+            }
+            state_dict.update(encodec_parameters)
+        vocoder.load_state_dict(state_dict)
+        vocoder = vocoder.eval().to(device)
+    elif vocoder_name == "bigvgan":
+        try:
+            import sys
+            sys.path.append('BigVGAN')
+            import bigvgan
+        except ImportError:
+            print("You need to follow the README to init submodule and change the BigVGAN source code.")
+        if is_local:
+            """download from https://huggingface.co/nvidia/bigvgan_v2_24khz_100band_256x/tree/main"""
+            vocoder = bigvgan.BigVGAN.from_pretrained(local_path, use_cuda_kernel=False)
+        else:
+            local_path = snapshot_download(repo_id="nvidia/bigvgan_v2_24khz_100band_256x", cache_dir=hf_cache_dir)
+            vocoder = bigvgan.BigVGAN.from_pretrained(local_path, use_cuda_kernel=False)
+
+        vocoder.remove_weight_norm()
+        vocoder = vocoder.eval().to(device)
+    return vocoder
+
+
+# load asr pipeline
+
+asr_pipe = None
+
+
+def initialize_asr_pipeline(device: str = device, dtype=None):
+    if dtype is None:
+        dtype = (
+            torch.float16 if "cuda" in device and torch.cuda.get_device_properties(device).major >= 6 else torch.float32
+        )
+    global asr_pipe
+    asr_pipe = pipeline(
+        "automatic-speech-recognition",
+        model="openai/whisper-large-v3-turbo",
+        torch_dtype=dtype,
+        device=device,
+    )
+
+
+# transcribe
+
+
+def transcribe(ref_audio, language=None):
+    global asr_pipe
+    if asr_pipe is None:
+        initialize_asr_pipeline(device=device)
+    return asr_pipe(
+        ref_audio,
+        chunk_length_s=30,
+        batch_size=128,
+        generate_kwargs={"task": "transcribe", "language": language} if language else {"task": "transcribe"},
+        return_timestamps=False,
+    )["text"].strip()
+
+
+# load model checkpoint for inference
+
+
+def load_checkpoint(model, ckpt_path, device: str, dtype=None, use_ema=True):
+    if dtype is None:
+        dtype = (
+            torch.float16 if "cuda" in device and torch.cuda.get_device_properties(device).major >= 6 else torch.float32
+        )
+    model = model.to(dtype)
+
+    ckpt_type = ckpt_path.split(".")[-1]
+    if ckpt_type == "safetensors":
+        from safetensors.torch import load_file
+
+        checkpoint = load_file(ckpt_path, device=device)
+    else:
+        checkpoint = torch.load(ckpt_path, map_location=device, weights_only=True)
+
+    if use_ema:
+        if ckpt_type == "safetensors":
+            checkpoint = {"ema_model_state_dict": checkpoint}
+        checkpoint["model_state_dict"] = {
+            k.replace("ema_model.", ""): v
+            for k, v in checkpoint["ema_model_state_dict"].items()
+            if k not in ["initted", "step"]
+        }
+
+        # patch for backward compatibility, 305e3ea
+        for key in ["mel_spec.mel_stft.mel_scale.fb", "mel_spec.mel_stft.spectrogram.window"]:
+            if key in checkpoint["model_state_dict"]:
+                del checkpoint["model_state_dict"][key]
+
+        model.load_state_dict(checkpoint["model_state_dict"])
+    else:
+        if ckpt_type == "safetensors":
+            checkpoint = {"model_state_dict": checkpoint}
+        model.load_state_dict(checkpoint["model_state_dict"])
+
+    del checkpoint
+    torch.cuda.empty_cache()
+
+    return model.to(device)
+
+
+# load model for inference
+
+
+def load_model(
+    model_cls,
+    model_cfg,
+    ckpt_path,
+    mel_spec_type=mel_spec_type,
+    vocab_file="",
+    ode_method=ode_method,
+    use_ema=True,
+    device=device,
+):
+    if vocab_file == "":
+        vocab_file = str(files("f5_tts").joinpath("infer/examples/vocab.txt"))
+    tokenizer = "custom"
+
+    print("\nvocab : ", vocab_file)
+    print("token : ", tokenizer)
+    print("model : ", ckpt_path, "\n")
+
+    vocab_char_map, vocab_size = get_tokenizer(vocab_file, tokenizer)
+    model = CFM(
+        transformer=model_cls(**model_cfg, text_num_embeds=vocab_size, mel_dim=n_mel_channels),
+        mel_spec_kwargs=dict(
+            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,
+        ),
+        odeint_kwargs=dict(
+            method=ode_method,
+        ),
+        vocab_char_map=vocab_char_map,
+    ).to(device)
+
+    dtype = torch.float32 if mel_spec_type == "bigvgan" else None
+    model = load_checkpoint(model, ckpt_path, device, dtype=dtype, use_ema=use_ema)
+
+    return model
+
+
+def remove_silence_edges(audio, silence_threshold=-42):
+    # Remove silence from the start
+    non_silent_start_idx = silence.detect_leading_silence(audio, silence_threshold=silence_threshold)
+    audio = audio[non_silent_start_idx:]
+
+    # Remove silence from the end
+    non_silent_end_duration = audio.duration_seconds
+    for ms in reversed(audio):
+        if ms.dBFS > silence_threshold:
+            break
+        non_silent_end_duration -= 0.001
+    trimmed_audio = audio[: int(non_silent_end_duration * 1000)]
+
+    return trimmed_audio
+
+
+# preprocess reference audio and text
+
+
+def preprocess_ref_audio_text(ref_audio_orig, ref_text, clip_short=True, show_info=print, device=device):
+    show_info("Converting audio...")
+    with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as f:
+        aseg = AudioSegment.from_file(ref_audio_orig)
+
+        if clip_short:
+            # 1. try to find long silence for clipping
+            non_silent_segs = silence.split_on_silence(
+                aseg, min_silence_len=1000, silence_thresh=-50, keep_silence=1000, seek_step=10
+            )
+            non_silent_wave = AudioSegment.silent(duration=0)
+            for non_silent_seg in non_silent_segs:
+                if len(non_silent_wave) > 6000 and len(non_silent_wave + non_silent_seg) > 15000:
+                    show_info("Audio is over 15s, clipping short. (1)")
+                    break
+                non_silent_wave += non_silent_seg
+
+            # 2. try to find short silence for clipping if 1. failed
+            if len(non_silent_wave) > 15000:
+                non_silent_segs = silence.split_on_silence(
+                    aseg, min_silence_len=100, silence_thresh=-40, keep_silence=1000, seek_step=10
+                )
+                non_silent_wave = AudioSegment.silent(duration=0)
+                for non_silent_seg in non_silent_segs:
+                    if len(non_silent_wave) > 6000 and len(non_silent_wave + non_silent_seg) > 15000:
+                        show_info("Audio is over 15s, clipping short. (2)")
+                        break
+                    non_silent_wave += non_silent_seg
+
+            aseg = non_silent_wave
+
+            # 3. if no proper silence found for clipping
+            if len(aseg) > 15000:
+                aseg = aseg[:15000]
+                show_info("Audio is over 15s, clipping short. (3)")
+
+        aseg = remove_silence_edges(aseg) + AudioSegment.silent(duration=50)
+        aseg.export(f.name, format="wav")
+        ref_audio = f.name
+
+    # Compute a hash of the reference audio file
+    with open(ref_audio, "rb") as audio_file:
+        audio_data = audio_file.read()
+        audio_hash = hashlib.md5(audio_data).hexdigest()
+
+    if not ref_text.strip():
+        global _ref_audio_cache
+        if audio_hash in _ref_audio_cache:
+            # Use cached asr transcription
+            show_info("Using cached reference text...")
+            ref_text = _ref_audio_cache[audio_hash]
+        else:
+            show_info("No reference text provided, transcribing reference audio...")
+            ref_text = transcribe(ref_audio)
+            # Cache the transcribed text (not caching custom ref_text, enabling users to do manual tweak)
+            _ref_audio_cache[audio_hash] = ref_text
+    else:
+        show_info("Using custom reference text...")
+
+    # Ensure ref_text ends with a proper sentence-ending punctuation
+    if not ref_text.endswith(". ") and not ref_text.endswith("。"):
+        if ref_text.endswith("."):
+            ref_text += " "
+        else:
+            ref_text += ". "
+
+    print("ref_text  ", ref_text)
+
+    return ref_audio, ref_text
+
+
+# infer process: chunk text -> infer batches [i.e. infer_batch_process()]
+
+
+def infer_process(
+    ref_audio,
+    ref_text,
+    gen_text,
+    model_obj,
+    vocoder,
+    mel_spec_type=mel_spec_type,
+    show_info=print,
+    progress=tqdm,
+    target_rms=target_rms,
+    cross_fade_duration=cross_fade_duration,
+    nfe_step=nfe_step,
+    cfg_strength=cfg_strength,
+    sway_sampling_coef=sway_sampling_coef,
+    speed=speed,
+    fix_duration=fix_duration,
+    device=device,
+):
+    # Split the input text into batches
+    audio, sr = torchaudio.load(ref_audio)
+    max_chars = int(len(ref_text.encode("utf-8")) / (audio.shape[-1] / sr) * (25 - audio.shape[-1] / sr))
+    print(f'{max_chars=}')
+    max_chars = 300  # 135
+
+    gen_text_batches = chunk_text(gen_text, max_chars=max_chars)
+    for i, gen_text in enumerate(gen_text_batches):
+        print(f"gen_text {i}", gen_text)
+
+    show_info(f"Generating audio in {len(gen_text_batches)} batches...")
+    return infer_batch_process(
+        (audio, sr),
+        ref_text,
+        gen_text_batches,
+        model_obj,
+        vocoder,
+        mel_spec_type=mel_spec_type,
+        progress=progress,
+        target_rms=target_rms,
+        cross_fade_duration=cross_fade_duration,
+        nfe_step=nfe_step,
+        cfg_strength=cfg_strength,
+        sway_sampling_coef=sway_sampling_coef,
+        speed=speed,
+        fix_duration=fix_duration,
+        device=device,
+    )
+
+
+# infer batches
+
+
+def infer_batch_process(
+    ref_audio,
+    ref_text,
+    gen_text_batches,
+    model_obj,
+    vocoder,
+    mel_spec_type="vocos",
+    progress=tqdm,
+    target_rms=0.1,
+    cross_fade_duration=0.15,
+    nfe_step=32,
+    cfg_strength=2.0,
+    sway_sampling_coef=-1,
+    speed=1,
+    fix_duration=None,
+    device=None,
+):
+    audio, sr = ref_audio
+    if audio.shape[0] > 1:
+        audio = torch.mean(audio, dim=0, keepdim=True)
+
+    rms = torch.sqrt(torch.mean(torch.square(audio)))
+    if rms < target_rms:
+        audio = audio * target_rms / rms
+    if sr != target_sample_rate:
+        resampler = torchaudio.transforms.Resample(sr, target_sample_rate)
+        audio = resampler(audio)
+    audio = audio.to(device)
+
+    generated_waves = []
+    spectrograms = []
+
+    if len(ref_text[-1].encode("utf-8")) == 1:
+        ref_text = ref_text + " "
+    for i, gen_text in enumerate(progress.tqdm(gen_text_batches)):
+        # Prepare the text
+        text_list = [ref_text + gen_text]
+        final_text_list = convert_char_to_pinyin(text_list)
+
+        ref_audio_len = audio.shape[-1] // hop_length
+        if fix_duration is not None:
+            duration = int(fix_duration * target_sample_rate / hop_length)
+        else:
+            # Calculate duration
+            ref_text_len = len(ref_text.encode("utf-8"))
+            gen_text_len = len(gen_text.encode("utf-8"))
+            duration = ref_audio_len + int(ref_audio_len / ref_text_len * gen_text_len / speed)
+
+        # inference
+        with torch.inference_mode():
+            generated, _ = model_obj.sample(
+                cond=audio,
+                text=final_text_list,
+                duration=duration,
+                steps=nfe_step,
+                cfg_strength=cfg_strength,
+                sway_sampling_coef=sway_sampling_coef,
+            )
+
+            generated = generated.to(torch.float32)
+            generated = generated[:, ref_audio_len:, :]
+            generated_mel_spec = generated.permute(0, 2, 1)
+            if mel_spec_type == "vocos":
+                generated_wave = vocoder.decode(generated_mel_spec)
+            elif mel_spec_type == "bigvgan":
+                generated_wave = vocoder(generated_mel_spec)
+            if rms < target_rms:
+                generated_wave = generated_wave * rms / target_rms
+
+            # wav -> numpy
+            generated_wave = generated_wave.squeeze().cpu().numpy()
+
+            generated_waves.append(generated_wave)
+            spectrograms.append(generated_mel_spec[0].cpu().numpy())
+
+    # Combine all generated waves with cross-fading
+    if cross_fade_duration <= 0:
+        # Simply concatenate
+        final_wave = np.concatenate(generated_waves)
+    else:
+        final_wave = generated_waves[0]
+        for i in range(1, len(generated_waves)):
+            prev_wave = final_wave
+            next_wave = generated_waves[i]
+
+            # Calculate cross-fade samples, ensuring it does not exceed wave lengths
+            cross_fade_samples = int(cross_fade_duration * target_sample_rate)
+            cross_fade_samples = min(cross_fade_samples, len(prev_wave), len(next_wave))
+
+            if cross_fade_samples <= 0:
+                # No overlap possible, concatenate
+                final_wave = np.concatenate([prev_wave, next_wave])
+                continue
+
+            # Overlapping parts
+            prev_overlap = prev_wave[-cross_fade_samples:]
+            next_overlap = next_wave[:cross_fade_samples]
+
+            # Fade out and fade in
+            fade_out = np.linspace(1, 0, cross_fade_samples)
+            fade_in = np.linspace(0, 1, cross_fade_samples)
+
+            # Cross-faded overlap
+            cross_faded_overlap = prev_overlap * fade_out + next_overlap * fade_in
+
+            # Combine
+            new_wave = np.concatenate(
+                [prev_wave[:-cross_fade_samples], cross_faded_overlap, next_wave[cross_fade_samples:]]
+            )
+
+            final_wave = new_wave
+
+    # Create a combined spectrogram
+    combined_spectrogram = np.concatenate(spectrograms, axis=1)
+
+    return final_wave, target_sample_rate, combined_spectrogram
+
+
+# remove silence from generated wav
+
+
+def remove_silence_for_generated_wav(filename):
+    aseg = AudioSegment.from_file(filename)
+    non_silent_segs = silence.split_on_silence(
+        aseg, min_silence_len=1000, silence_thresh=-50, keep_silence=500, seek_step=10
+    )
+    non_silent_wave = AudioSegment.silent(duration=0)
+    for non_silent_seg in non_silent_segs:
+        non_silent_wave += non_silent_seg
+    aseg = non_silent_wave
+    aseg.export(filename, format="wav")
+
+
+# save spectrogram
+
+
+def save_spectrogram(spectrogram, path):
+    plt.figure(figsize=(12, 4))
+    plt.imshow(spectrogram, origin="lower", aspect="auto")
+    plt.colorbar()
+    plt.savefig(path)
+    plt.close()

model.py

@@ -0,0 +1,285 @@
+"""
+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 torch import nn
+from torch.nn.utils.rnn import pad_sequence
+from torchdiffeq import odeint
+
+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,
+)
+
+
+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

model_modules.py

@@ -0,0 +1,658 @@
+"""
+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

model_utils.py

@@ -0,0 +1,187 @@
+from __future__ import annotations
+
+import os
+import random
+from collections import defaultdict
+from importlib.resources import files
+
+import torch
+from torch.nn.utils.rnn import pad_sequence
+
+import jieba
+from pypinyin import lazy_pinyin, Style
+
+
+# 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) -> 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) -> float["b d"]:  # noqa: F722
+    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
+
+
+def convert_char_to_pinyin(text_list, polyphone=True):
+    final_text_list = []
+    god_knows_why_en_testset_contains_zh_quote = str.maketrans(
+        {"“": '"', "”": '"', "‘": "'", "’": "'"}
+    )  # in case librispeech (orig no-pc) test-clean
+    custom_trans = str.maketrans({";": ","})  # add custom trans here, to address oov
+    for text in text_list:
+        char_list = []
+        text = text.translate(god_knows_why_en_testset_contains_zh_quote)
+        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 chinese characters
+                seg = lazy_pinyin(seg, style=Style.TONE3, tone_sandhi=True)
+                for c in seg:
+                    if c not in "。，、；：？！《》【】—…":
+                        char_list.append(" ")
+                    char_list.append(c)
+            else:  # if mixed chinese characters, alphabets and symbols
+                for c in seg:
+                    if ord(c) < 256:
+                        char_list.extend(c)
+                    elif '\u0400' <= c <= '\u04FF':  # Cyrillic Unicode block
+                        char_list.extend(c)
+                    else:
+                        if c not in "。，、；：？！《》【】—…":
+                            char_list.append(" ")
+                            char_list.extend(lazy_pinyin(c, style=Style.TONE3, tone_sandhi=True))
+                        else:  # if is zh punc
+                            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

requirements.txt

@@ -0,0 +1,26 @@
+accelerate>=0.33.0
+bitsandbytes>0.37.0
+cached_path
+click
+datasets
+ema_pytorch>=0.5.2
+gradio>=3.45.2
+hydra-core>=1.3.0
+jieba
+librosa
+matplotlib
+numpy<=1.26.4
+pydub
+pypinyin
+safetensors
+soundfile
+tomli
+torch>=2.0.0
+torchaudio>=2.0.0
+torchdiffeq
+tqdm>=4.65.0
+transformers
+transformers_stream_generator
+vocos
+wandb
+x_transformers>=1.31.14