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https://github.com/malarinv/tacotron2
synced 2026-03-08 01:32:35 +00:00
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| Author | SHA1 | Date | |
|---|---|---|---|
| 6d3679d760 | |||
| a851e80db2 | |||
| cb0c8ddd06 | |||
| 42a85d177e | |||
| 5efb1e2758 |
19
setup.py
19
setup.py
@@ -15,12 +15,19 @@ requirements = [
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"numpy~=1.16.4",
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"inflect==0.2.5",
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"librosa==0.6.0",
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"scipy~=1.3.0",
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"scipy~=1.3",
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"Unidecode==1.0.22",
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"torch~=1.1.0",
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"PyAudio==0.2.11"
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]
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extra_requirements = {
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"playback": ["PyAudio==0.2.11"],
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"server": [
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"google-cloud-texttospeech==1.0.1",
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"rpyc==4.1.4",
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],
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}
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setup_requirements = ["pytest-runner"]
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test_requirements = ["pytest"]
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@@ -44,6 +51,7 @@ setup(
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],
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description="Taco2 TTS package.",
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install_requires=requirements,
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extras_require=extra_requirements,
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long_description=readme + "\n\n" + history,
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include_package_data=True,
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keywords="tacotron2 tts",
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@@ -55,5 +63,10 @@ setup(
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url="https://github.com/malarinv/tacotron2",
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version="0.3.0",
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zip_safe=False,
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entry_points={"console_scripts": ("tts_debug = taco2.tts:main",)},
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entry_points={
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"console_scripts": (
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"tts_debug = taco2.tts:main",
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"tts_rpyc_server = taco2.server.__main__:main",
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)
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},
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)
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0
taco2/server/__init__.py
Normal file
0
taco2/server/__init__.py
Normal file
48
taco2/server/__main__.py
Normal file
48
taco2/server/__main__.py
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@@ -0,0 +1,48 @@
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import os
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import logging
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import rpyc
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from rpyc.utils.server import ThreadedServer
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from .backend import TTSSynthesizer
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tts_backend = os.environ.get("TTS_BACKEND", "taco2")
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tts_synthesizer = TTSSynthesizer(backend=tts_backend)
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class TTSService(rpyc.Service):
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def on_connect(self, conn):
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# code that runs when a connection is created
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# (to init the service, if needed)
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pass
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def on_disconnect(self, conn):
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# code that runs after the connection has already closed
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# (to finalize the service, if needed)
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pass
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def exposed_synth_speech(self, utterance: str): # this is an exposed method
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speech_audio = tts_synthesizer.synth_speech(utterance)
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return speech_audio
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def exposed_synth_speech_cb(
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self, utterance: str, respond
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): # this is an exposed method
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speech_audio = tts_synthesizer.synth_speech(utterance)
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respond(speech_audio)
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def main():
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logging.basicConfig(
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level=logging.INFO,
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format="%(asctime)s - %(name)s - %(levelname)s - %(message)s",
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)
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port = int(os.environ.get("TTS_RPYC_PORT", "7754"))
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logging.info("starting tts server...")
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t = ThreadedServer(TTSService, port=port)
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t.start()
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if __name__ == "__main__":
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main()
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45
taco2/server/backend.py
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45
taco2/server/backend.py
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@@ -0,0 +1,45 @@
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import os
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from google.cloud import texttospeech
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from ..tts import TTSModel
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tts_model_weights = os.environ.get(
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"TTS_MODELS", "models/tacotron2_statedict.pt,models/waveglow_256channels.pt"
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)
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tts_creds = os.environ.get(
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"GOOGLE_APPLICATION_CREDENTIALS", "/code/config/gre2e/keys/gre2e_gcp.json"
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)
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taco2, wav_glow = tts_model_weights.split(",", 1)
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class TTSSynthesizer(object):
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"""docstring for TTSSynthesizer."""
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def __init__(self, backend="taco2"):
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super(TTSSynthesizer, self).__init__()
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if backend == "taco2":
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tts_model = TTSModel(f"{taco2}", f"{wav_glow}") # Loads the models
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self.synth_speech = tts_model.synth_speech
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elif backend == "gcp":
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client = texttospeech.TextToSpeechClient()
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# Build the voice request, select the language code ("en-US") and the ssml
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# voice gender ("neutral")
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voice = texttospeech.types.VoiceSelectionParams(language_code="en-US")
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# Select the type of audio file you want returned
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audio_config = texttospeech.types.AudioConfig(
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audio_encoding=texttospeech.enums.AudioEncoding.LINEAR16
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)
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# Perform the text-to-speech request on the text input with the selected
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# voice parameters and audio file type
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def gcp_synthesize(speech_text):
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synthesis_input = texttospeech.types.SynthesisInput(text=speech_text)
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response = client.synthesize_speech(
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synthesis_input, voice, audio_config
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)
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return response.audio_content
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self.synth_speech = gcp_synthesize
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55
taco2/tts.py
55
taco2/tts.py
@@ -3,9 +3,9 @@
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import numpy as np
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import torch
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import pyaudio
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import klepto
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import argparse
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import warnings
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from pathlib import Path
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from .model import Tacotron2
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from glow import WaveGlow
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@@ -17,7 +17,7 @@ from .audio_processing import griffin_lim, postprocess_audio
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OUTPUT_SAMPLE_RATE = 22050
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GL_ITERS = 30
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VOCODER_MODEL = "wavglow"
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VOCODER_WAVEGLOW, VOCODER_GL = "wavglow", "gl"
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# config from
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# https://github.com/NVIDIA/waveglow/blob/master/config.json
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@@ -74,11 +74,11 @@ class TTSModel(object):
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self.waveglow, n_mel_channels=hparams.n_mel_channels
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)
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self.synth_speech = klepto.safe.inf_cache(cache=self.k_cache)(
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self.synth_speech
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self._synth_speech
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)
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else:
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self.synth_speech = klepto.safe.inf_cache(cache=self.k_cache)(
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self.synth_speech_fast
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self._synth_speech_fast
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)
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self.taco_stft = TacotronSTFT(
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hparams.filter_length,
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@@ -89,7 +89,7 @@ class TTSModel(object):
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mel_fmax=4000,
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)
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def generate_mel_postnet(self, text):
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def _generate_mel_postnet(self, text):
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sequence = np.array(text_to_sequence(text, ["english_cleaners"]))[None, :]
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if torch.cuda.is_available():
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sequence = torch.autograd.Variable(torch.from_numpy(sequence)).cuda().long()
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@@ -102,14 +102,14 @@ class TTSModel(object):
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return mel_outputs_postnet
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def synth_speech_array(self, text, vocoder):
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mel_outputs_postnet = self.generate_mel_postnet(text)
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mel_outputs_postnet = self._generate_mel_postnet(text)
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if vocoder == "wavglow":
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if vocoder == VOCODER_WAVEGLOW:
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with torch.no_grad():
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audio_t = self.waveglow.infer(mel_outputs_postnet, sigma=0.666)
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audio_t = self.denoiser(audio_t, 0.1)[0]
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audio = audio_t[0].data
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elif vocoder == "gl":
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elif vocoder == VOCODER_GL:
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mel_decompress = self.taco_stft.spectral_de_normalize(mel_outputs_postnet)
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mel_decompress = mel_decompress.transpose(1, 2).data.cpu()
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spec_from_mel_scaling = 1000
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@@ -122,7 +122,7 @@ class TTSModel(object):
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audio = griffin_lim(
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torch.autograd.Variable(spec_from_mel[:, :, :-1]),
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self.taco_stft.stft_fn,
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60,
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GL_ITERS,
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)
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audio = audio.squeeze()
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else:
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@@ -130,40 +130,37 @@ class TTSModel(object):
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audio = audio.cpu().numpy()
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return audio
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def synth_speech(
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def _synth_speech(
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self, text, speed: float = 1.0, sample_rate: int = OUTPUT_SAMPLE_RATE
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):
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audio = self.synth_speech_array(text, VOCODER_MODEL)
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audio = self.synth_speech_array(text, VOCODER_WAVEGLOW)
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return postprocess_audio(
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audio, src_rate=self.hparams.sampling_rate, dst_rate=sample_rate, tempo=speed
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audio,
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src_rate=self.hparams.sampling_rate,
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dst_rate=sample_rate,
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tempo=speed,
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)
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def synth_speech_fast(
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def _synth_speech_fast(
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self, text, speed: float = 1.0, sample_rate: int = OUTPUT_SAMPLE_RATE
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):
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mel_outputs_postnet = self.generate_mel_postnet(text)
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mel_decompress = self.taco_stft.spectral_de_normalize(mel_outputs_postnet)
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mel_decompress = mel_decompress.transpose(1, 2).data.cpu()
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spec_from_mel_scaling = 1000
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spec_from_mel = torch.mm(mel_decompress[0], self.taco_stft.mel_basis)
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spec_from_mel = spec_from_mel.transpose(0, 1).unsqueeze(0)
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spec_from_mel = spec_from_mel * spec_from_mel_scaling
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audio = griffin_lim(
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torch.autograd.Variable(spec_from_mel[:, :, :-1]),
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self.taco_stft.stft_fn,
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GL_ITERS,
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)
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audio = audio.squeeze()
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audio = audio.cpu().numpy()
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audio = self.synth_speech_array(text, VOCODER_GL)
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return postprocess_audio(
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audio, tempo=speed, src_rate=self.hparams.sampling_rate, dst_rate=sample_rate
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audio,
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tempo=speed,
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src_rate=self.hparams.sampling_rate,
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dst_rate=sample_rate,
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)
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def player_gen():
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try:
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import pyaudio
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except ModuleNotFoundError:
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warnings.warn("module 'pyaudio' is not installed requried for playback")
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return
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audio_interface = pyaudio.PyAudio()
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_audio_stream = audio_interface.open(
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format=pyaudio.paInt16, channels=1, rate=OUTPUT_SAMPLE_RATE, output=True
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