appPlayer2.py

import os
os.environ.setdefault("OMP_NUM_THREADS", "1")
os.environ.setdefault("MKL_NUM_THREADS", "1")
os.environ.setdefault("NUMEXPR_NUM_THREADS", "1")

import sys
import re
import time
import json
import base64
import hashlib
import tempfile
import subprocess
from typing import Iterator, Iterable, Optional, Tuple, Any, List
from dataclasses import dataclass
import pathlib

import spaces
import gradio as gr
import torch
import numpy as np
from huggingface_hub import hf_hub_download
from scipy.io.wavfile import write

# ---------------------------------------------------------
# 1) coqui-ai-TTS fork
# ---------------------------------------------------------
REPO_URL = "https://github.com/tuteishygpt/coqui-ai-TTS.git"
REPO_DIR = "coqui-ai-TTS"
if not os.path.exists(REPO_DIR): subprocess.run(["git", "clone", REPO_URL, REPO_DIR], check=True)
repo_root = os.path.abspath(REPO_DIR)
if repo_root not in sys.path: sys.path.insert(0, repo_root)

from TTS.tts.configs.xtts_config import XttsConfig
from TTS.tts.models.xtts import Xtts
from TTS.tts.layers.xtts.tokenizer import VoiceBpeTokenizer, split_sentence

# ---------------------------------------------------------
# 2) мадэльныя файлы
# ---------------------------------------------------------
repo_id = "archivartaunik/BE_XTTS_V2_10ep250k"
model_dir = pathlib.Path("./model")
model_dir.mkdir(exist_ok=True)
for fname in ("model.pth", "config.json", "vocab.json", "voice.wav"):
    if not (model_dir / fname).exists(): hf_hub_download(repo_id, filename=fname, local_dir=model_dir)

# ---------------------------------------------------------
# 3) загрузка мадэлі
# ---------------------------------------------------------
config = XttsConfig()
config.load_json(str(model_dir / "config.json"))
XTTS_MODEL = Xtts.init_from_config(config)
XTTS_MODEL.load_checkpoint(config, checkpoint_path=str(model_dir / "model.pth"), vocab_path=str(model_dir / "vocab.json"), use_deepspeed=False)

device = "cuda:0" if torch.cuda.is_available() else "cpu"
if device.startswith("cuda"):
    torch.cuda.manual_seed(0)
    torch.backends.cuda.matmul.allow_tf32 = True
    torch.backends.cudnn.allow_tf32 = True
XTTS_MODEL.to(device)
sampling_rate = int(XTTS_MODEL.config.audio["sample_rate"])

# =========================================================
# 4) Streaming-канфіг і дапаўненні
# =========================================================
INITIAL_MIN_BUFFER_S = 0.40
MIN_BUFFER_S = 0.15
FADE_S = 0.005
ENABLE_TEXT_SPLITTING = True

def _to_np_audio(x) -> np.ndarray:
    if isinstance(x, dict) and "wav" in x: x = x["wav"]
    if isinstance(x, torch.Tensor): x = x.detach().cpu().float().contiguous().view(-1).numpy()
    x = np.asarray(x, dtype=np.float32)
    return x.reshape(-1)

# ---------------------------------------------------------
# 5) пастаянны кэш латэнтаў
# ---------------------------------------------------------
PERSIST_LATENTS_DIR = pathlib.Path("./latents_cache")
PERSIST_LATENTS_DIR.mkdir(parents=True, exist_ok=True)
@dataclass(frozen=True)
class LatentsMeta: model_id: str; gpt_cond_len: int; max_ref_len: int; sound_norm_refs: bool
LATENT_CACHE: dict[str, Tuple[torch.Tensor, torch.Tensor]] = {}
GPU_LATENT_CACHE: dict[Tuple[str, str], Tuple[torch.Tensor, torch.Tensor]] = {}
default_voice_file = str(model_dir / "voice.wav")

def _latents_key(path: str | None, meta: LatentsMeta) -> str:
    base = f"{os.path.abspath(path)}:{os.path.getmtime(path)}:{os.path.getsize(path)}" if path and os.path.exists(path) else "default_voice"
    return hashlib.md5((base + "|" + json.dumps(meta.__dict__, sort_keys=True)).encode("utf-8")).hexdigest()

def _latents_for(path: str | None, *, to_device: Optional[str] = None) -> Tuple[torch.Tensor, torch.Tensor]:
    meta = LatentsMeta(model_id=repo_id, gpt_cond_len=XTTS_MODEL.config.gpt_cond_len, max_ref_len=XTTS_MODEL.config.max_ref_len, sound_norm_refs=XTTS_MODEL.config.sound_norm_refs)
    key = _latents_key(path, meta)
    g, s = LATENT_CACHE.get(key) or (None, None)
    if g is None:
        disk_path = PERSIST_LATENTS_DIR / f"{key}.pt"
        if disk_path.exists():
            data = torch.load(disk_path, map_location="cpu"); g, s = data["gpt_cond_latent"], data["speaker_embedding"]
        else:
            with torch.inference_mode(): g_cpu, s_cpu = XTTS_MODEL.get_conditioning_latents(audio_path=path)
            g, s = g_cpu.cpu(), s_cpu.cpu()
            torch.save({"gpt_cond_latent": g, "speaker_embedding": s}, disk_path)
        LATENT_CACHE[key] = (g, s)
    if to_device:
        dev_key = (key, to_device)
        if dev_key in GPU_LATENT_CACHE: return GPU_LATENT_CACHE[dev_key]
        g, s = g.to(to_device, non_blocking=True), s.to(to_device, non_blocking=True)
        GPU_LATENT_CACHE[dev_key] = (g, s)
    return g, s

try: _latents_for(default_voice_file, to_device=device)
except Exception as e: print(f"Warning: Could not pre-cache default voice: {e}")

# ---------------------------------------------------------
# 7) Дапаможныя функцыі для стрыму
# ---------------------------------------------------------
def _seconds_to_samples(sec: float, sr: int) -> int: return max(1, int(sec * sr))
def _crossfade_concat(chunks: List[np.ndarray], sr: int, fade_s: float) -> np.ndarray:
    if not chunks: return np.array([], dtype=np.float32)
    result = chunks[0]
    for i in range(1, len(chunks)):
        b = chunks[i]
        fade_n = min(_seconds_to_samples(fade_s, sr), result.size, b.size)
        if fade_n <= 1: result = np.concatenate([result, b]); continue
        fade_out, fade_in = np.linspace(1.0, 0.0, fade_n, dtype=np.float32), np.linspace(0.0, 1.0, fade_n, dtype=np.float32)
        tail = (result[-fade_n:] * fade_out) + (b[:fade_n] * fade_in)
        result = np.concatenate([result[:-fade_n], tail, b[fade_n:]])
    return result

def _chunker(chunks: Iterable[np.ndarray], sr: int, initial_target_s: float, target_s: float) -> Iterator[np.ndarray]:
    is_first, target_samples = True, _seconds_to_samples(initial_target_s, sr)
    buffer = np.array([], dtype=np.float32)
    for c_np in map(_to_np_audio, chunks):
        if c_np.size == 0: continue
        buffer = np.concatenate([buffer, c_np])
        if buffer.size >= target_samples:
            yield buffer
            buffer = np.array([], dtype=np.float32)
            if is_first: is_first = False; target_samples = _seconds_to_samples(target_s, sr)
    if buffer.size > 0: yield buffer

def _pcm_f32_to_b64(x: np.ndarray) -> str: return base64.b64encode(x.tobytes()).decode("ascii")

def _split_text_smart(text: str, lang: str, limit: int) -> List[str]:
    try: sentences = split_sentence(text, lang=lang)
    except Exception: sentences = [text]
    chunks, current_chunk = [], ""
    for sentence in sentences:
        if len(current_chunk) + len(sentence) + 1 > limit and current_chunk:
            chunks.append(current_chunk); current_chunk = ""
        current_chunk = (current_chunk + " " + sentence).strip()
    if current_chunk: chunks.append(current_chunk)
    final_chunks = []
    for chunk in chunks:
        if len(chunk) > limit: final_chunks.extend(chunk[i:i+limit] for i in range(0, len(chunk), limit))
        else: final_chunks.append(chunk)
    return [c.strip() for c in final_chunks if c.strip()]

# ---------------------------------------------------------
# 8) TTS — асноўная функцыя
# ---------------------------------------------------------
@spaces.GPU(duration=120)
def text_to_speech(text_input, speaker_audio, initial_buffer_s, subsequent_buffer_s):
    t_start_req = time.perf_counter()
    if not text_input or not str(text_input).strip(): raise gr.Error("Увядзі хоць нейкі тэкст 🙂")
    
    t_lat_0 = time.perf_counter()
    gpt_cond_latent, speaker_embedding = _latents_for(speaker_audio or default_voice_file, to_device=device)
    t_lat_1 = time.perf_counter()

    t_split_0 = time.perf_counter()
    char_limit = XTTS_MODEL.tokenizer.char_limits.get("be", 250)
    texts = _split_text_smart(str(text_input).strip(), "be", char_limit) if ENABLE_TEXT_SPLITTING else [str(text_input).strip()]
    t_split_1 = time.perf_counter()
    
    server_metrics = { "latents_s": t_lat_1 - t_lat_0, "text_split_s": t_split_1 - t_split_0, "initial_buffer_s": initial_buffer_s, "subsequent_buffer_s": subsequent_buffer_s }
    yield ("", None, None, json.dumps(server_metrics))
    
    full_audio_chunks, first_chunk_sent = [], False
    t_gen_start = time.perf_counter()
    
    with torch.inference_mode(), torch.autocast(device_type="cuda", dtype=torch.float16, enabled=device.startswith("cuda")):
        all_chunks_iterator = (
            _to_np_audio(chunk) for part in texts for chunk in XTTS_MODEL.inference_stream(
                text=part, language="be", gpt_cond_latent=gpt_cond_latent, speaker_embedding=speaker_embedding,
                temperature=0.2, length_penalty=1.0, repetition_penalty=10.0, top_k=20, top_p=0.85
            )
        )

        for audio_chunk in _chunker(all_chunks_iterator, sampling_rate, initial_buffer_s, subsequent_buffer_s):
            if not first_chunk_sent:
                t_first_chunk_ready = time.perf_counter()
                server_metrics["gen_init_to_first_chunk_s"] = t_first_chunk_ready - t_gen_start
                server_metrics["until_first_chunk_total_s"] = t_first_chunk_ready - t_start_req
                yield (_pcm_f32_to_b64(audio_chunk), None, None, json.dumps(server_metrics))
                first_chunk_sent = True
            else:
                yield (_pcm_f32_to_b64(audio_chunk), None, None, None)
            full_audio_chunks.append(audio_chunk)

    if not full_audio_chunks:
        yield ("__STOP__", None, None, json.dumps(server_metrics)); return

    t_write_0 = time.perf_counter()
    full_audio = _crossfade_concat(full_audio_chunks, sampling_rate, FADE_S)
    with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as tmp:
        write(tmp.name, sampling_rate, full_audio)
    server_metrics["file_write_s"] = time.perf_counter() - t_write_0
    yield ("__STOP__", tmp.name, tmp.name, json.dumps(server_metrics))

# ---------------------------------------------------------
# 9) UI
# ---------------------------------------------------------
examples = [["Прывітанне! Гэта праверка жывога струменя беларускага TTS.", None, INITIAL_MIN_BUFFER_S, MIN_BUFFER_S]]
with gr.Blocks() as demo:
    gr.Markdown("## Belarusian TTS — Streaming (стабільны старт) + фінальны файл")
    with gr.Row():
        inp_text = gr.Textbox(lines=5, label="Тэкст на беларускай мове")
        inp_voice = gr.Audio(type="filepath", label="Прыклад голасу (6–10 сек)")
    with gr.Accordion("Дадатковыя налады стрымінгу", open=True):
        initial_buffer_slider = gr.Slider(minimum=0.1, maximum=1.5, value=INITIAL_MIN_BUFFER_S, step=0.05, label="Пачатковы буфер (с)")
        subsequent_buffer_slider = gr.Slider(minimum=0.05, maximum=0.5, value=MIN_BUFFER_S, step=0.01, label="Наступны буфер (с)")
    with gr.Row(): run_btn = gr.Button("Згенераваць"); gr.Markdown(f"**Sample rate:** {sampling_rate} Hz")
    log_panel = gr.HTML(value='<div id="wa-log" style="font-family:monospace;font-size:12px;white-space:pre-line">[лог пусты]</div>', label="Лагі плэера")
    stream_pipe, log_pipe, final_file, final_audio = gr.Textbox(visible=False), gr.Textbox(visible=False), gr.File(label="Згенераваны WAV"), gr.Audio(label="Фінальнае аўдыя", type="filepath")

    # <--- ВЫПРАЎЛЕННЕ: Увесь JS код вынесены ў адну зменную для надзейнасці
    # І ён вяртае функцыю, як таго патрабуе Gradio
    JS_CODE = f"""
function() {{
  const sampleRate = {sampling_rate};
  // Ініцыялізацыя або скід стану плэера
  function initOrResetPlayer() {{
    if (window.__wa) {{ window.__wa.reset(); return; }}
    const AC = window.AudioContext || window.webkitAudioContext;
    if (!AC) {{ console.error("AudioContext is not supported."); return; }}
    const ctx = new AC({{ sampleRate }});
    const node = ctx.createScriptProcessor(4096, 1, 1);
    let queue = [], playing = false, eos = false;
    let meta = {{ t_click_ms: performance.now(), chunk_durations: [] }};
    
    node.onaudioprocess = (e) => {{
      const out = e.outputBuffer.getChannelData(0); let i = 0;
      while (i < out.length) {{
        if (queue.length === 0 || !playing) {{ out[i++] = 0.0; continue; }}
        let cur = queue[0];
        const take = Math.min(cur.length, out.length - i);
        if (meta.t_first_audio_ms === null) {{ meta.t_first_audio_ms = performance.now(); logUpdate(); }}
        out.set(cur.subarray(0, take), i); i += take;
        if (take === cur.length) queue.shift(); else queue[0] = cur.subarray(take);
      }}
      if (eos && queue.length === 0 && playing) {{ playing = false; logUpdate(); }}
    }};
    node.connect(ctx.destination);

    function fmtS(x) {{ return x === null || x === undefined ? "n/a" : x.toFixed(3) + " s"; }}
    function logUpdate() {{
      const el = document.getElementById('wa-log'); if (!el) return;
      const s = meta.server || {{}}; const lines = ["Клік (Згенераваць): 0.000 s"];
      if (meta.t_first_push_ms) {{
        lines.push("Першы чанк прыйшоў:   " + fmtS((meta.t_first_push_ms - meta.t_click_ms) / 1000));
        if (meta.t_first_audio_ms) {{
          lines.push("Пачатак прайгравання: " + fmtS((meta.t_first_audio_ms - meta.t_click_ms) / 1000));
          lines.push("Затрымка (чанк→аўдыя): " + fmtS((meta.t_first_audio_ms - meta.t_first_push_ms) / 1000));
        }}
      }}
      lines.push("\\n— Налады стрыму —", "Пачатковы буфер (запыт):  " + fmtS(s.initial_buffer_s), "Наступны буфер (запыт):   " + fmtS(s.subsequent_buffer_s));
      if (meta.chunk_durations.length > 0) {{ lines.push("Працягласць 1-га чанка:    " + meta.chunk_durations[0] + " s", "Атрымана чанкаў:          " + meta.chunk_durations.length); }}
      lines.push("\\n— Серверныя метрыкі —", "Latents (умоўны голас):  " + fmtS(s.latents_s), "Падзел тэксту:           " + fmtS(s.text_split_s), "Ініт→1-ы чанк:           " + fmtS(s.gen_init_to_first_chunk_s), "Усё да 1-га чанка:       " + fmtS(s.until_first_chunk_total_s));
      if (meta.t_first_push_ms && s.until_first_chunk_total_s) {{ lines.push("\\nАцэнка чаргі ZeroGPU + сеткі: " + fmtS(Math.max(0, (meta.t_first_push_ms - meta.t_click_ms) / 1000 - s.until_first_chunk_total_s))); }}
      lines.push("\\nСтатус стриму: " + (playing ? "playing" : "stopped"));
      el.innerHTML = lines.join("\\n");
    }}

    window.__wa = {{
      push: (b64) => {{
        if (!b64 || b64 === "__STOP__") {{ eos = true; logUpdate(); return; }}
        const bin = atob(b64); const buf = new ArrayBuffer(bin.length); const view = new Uint8Array(buf);
        for (let i=0; i<bin.length; i++) view[i] = bin.charCodeAt(i);
        const f32 = new Float32Array(buf);
        if (meta.chunk_durations.length === 0 && f32.length > 0) meta.t_first_push_ms = performance.now();
        meta.chunk_durations.push((f32.length / ctx.sampleRate).toFixed(3));
        queue.push(f32);
        if (!playing && queue.length > 0) {{ playing = true; if(ctx.state === "suspended") ctx.resume(); }}
        logUpdate();
      }},
      update_server_metrics: (js) => {{ if(js) meta.server = JSON.parse(js); logUpdate(); }},
      reset: () => {{
        playing = false; eos = false; queue.length = 0;
        meta = {{ t_click_ms: performance.now(), chunk_durations: [], server: null }}; logUpdate();
      }},
    }};
  }}
  initOrResetPlayer();
}}
"""
    # <--- ВЫПРАЎЛЕННЕ: Простыя і надзейныя апрацоўшчыкі падзей
    run_btn.click(fn=None, js=JS_CODE)
    run_btn.click(
        fn=text_to_speech,
        inputs=[inp_text, inp_voice, initial_buffer_slider, subsequent_buffer_slider],
        outputs=[stream_pipe, final_file, final_audio, log_pipe],
    )
    stream_pipe.change(fn=None, inputs=[stream_pipe], js="(b64) => { if(window.__wa) window.__wa.push(b64); }")
    log_pipe.change(fn=None, inputs=[log_pipe], js="(js) => { if(window.__wa) window.__wa.update_server_metrics(js); }")
    
    gr.Examples(examples=examples, inputs=[inp_text, inp_voice, initial_buffer_slider, subsequent_buffer_slider], cache_examples=False)

if __name__ == "__main__":
    demo.launch()