Allow for playing the audio in a streaming way.
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@ -2,16 +2,16 @@
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# requires-python = ">=3.12"
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# dependencies = [
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# "huggingface_hub",
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# "moshi_mlx",
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# "moshi_mlx>=0.2.8",
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# "numpy",
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# "sounddevice",
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# ]
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# ///
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import argparse
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from dataclasses import dataclass
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import json
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from pathlib import Path
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import queue
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import time
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import numpy as np
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@ -104,6 +104,7 @@ def main():
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cfg_is_no_prefix = True
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mimi = tts_model.mimi
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log("info", f"reading input from {args.inp}")
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with open(args.inp, "r") as fobj:
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text_to_tts = fobj.read().strip()
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@ -114,39 +115,56 @@ def main():
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voices = []
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all_attributes = [tts_model.make_condition_attributes(voices, cfg_coef_conditioning)]
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begin = time.time()
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result = tts_model.generate(
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all_entries, all_attributes,
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cfg_is_no_prefix=cfg_is_no_prefix, cfg_is_no_text=cfg_is_no_text)
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frames = mx.concat(result.frames, axis=-1)
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total_duration = frames.shape[0] * frames.shape[-1] / mimi.frame_rate
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time_taken = time.time() - begin
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total_speed = total_duration / time_taken
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log("info", f"[LM] took {time_taken:.2f}s, total speed {total_speed:.2f}x")
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wav_frames = queue.Queue()
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def _on_audio_hook(audio_tokens):
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if (audio_tokens == -1).any():
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return
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_pcm = tts_model.mimi.decode_step(audio_tokens[None, :, None])
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_pcm = np.array(mx.clip(_pcm[0, 0], -1, 1))
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wav_frames.put_nowait(_pcm)
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def run():
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log("info", "starting the inference loop")
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begin = time.time()
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result = tts_model.generate(
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all_entries,
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all_attributes,
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cfg_is_no_prefix=cfg_is_no_prefix,
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cfg_is_no_text=cfg_is_no_text,
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on_audio_hook=_on_audio_hook,
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)
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frames = mx.concat(result.frames, axis=-1)
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total_duration = frames.shape[0] * frames.shape[-1] / mimi.frame_rate
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time_taken = time.time() - begin
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total_speed = total_duration / time_taken
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log("info", f"[LM] took {time_taken:.2f}s, total speed {total_speed:.2f}x")
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return result
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wav_frames = []
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for frame in result.frames:
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# We are processing frames one by one, although we could group them to improve speed.
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_pcm = tts_model.mimi.decode_step(frame)
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wav_frames.append(_pcm)
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if args.out == "-":
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cnt = [0]
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def audio_callback(outdata, _a, _b, _c):
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if cnt[0] < len(wav_frames):
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outdata[:, 0] = wav_frames[cnt[0]][0, 0]
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cnt[0] += 1
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else:
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try:
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pcm_data = wav_frames.get(block=False)
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outdata[:, 0] = pcm_data
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except queue.Empty:
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outdata[:] = 0
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with sd.OutputStream(samplerate=mimi.sample_rate,
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blocksize=1920,
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channels=1,
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callback=audio_callback):
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time.sleep(10)
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run()
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time.sleep(3)
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while True:
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if wav_frames.qsize() == 0:
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break
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time.sleep(1)
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else:
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wavs = mx.concat(wav_frames, axis=-1)
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end_step = result.end_steps[0]
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wav_length = int((mimi.sample_rate * (end_step + tts_model.final_padding) / mimi.frame_rate))
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wav = np.array(mx.clip(wavs[0, :, :wav_length], -1, 1))
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frames = []
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while True:
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try:
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frames.append(wav_frames.get_nowait())
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except queue.Empty:
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break
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wav = np.concat(frames, -1)
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sphn.write_wav(args.out, wav, mimi.sample_rate)
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