Spaces:
Running
on
T4
Running
on
T4
gorkemgoknar
commited on
Commit
•
156829e
1
Parent(s):
f7c2b84
Use xtts-streaming for generation
Browse files
app.py
CHANGED
@@ -3,16 +3,17 @@ import io, os, stat
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import subprocess
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import random
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from zipfile import ZipFile
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import uuid
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import time
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import torch
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import torchaudio
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# By using XTTS you agree to CPML license https://coqui.ai/cpml
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os.environ["COQUI_TOS_AGREED"] = "1"
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# langid is used to detect language for longer text
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# Most users expect text to be their own language, there is checkbox to disable it
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import langid
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import base64
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import csv
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from io import StringIO
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@@ -39,12 +40,13 @@ repo_id = "coqui/xtts"
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print("Export newer ffmpeg binary for denoise filter")
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ZipFile("ffmpeg.zip").extractall()
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print("Make ffmpeg binary executable")
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st = os.stat(
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os.chmod(
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# This will trigger downloading model
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print("Downloading if not downloaded Coqui XTTS V1.1")
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from TTS.utils.manage import ModelManager
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model_name = "tts_models/multilingual/multi-dataset/xtts_v1.1"
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ModelManager().download_model(model_name)
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model_path = os.path.join(get_user_data_dir("tts"), model_name.replace("/", "--"))
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@@ -63,156 +65,188 @@ model.load_checkpoint(
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checkpoint_path=os.path.join(model_path, "model.pth"),
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vocab_path=os.path.join(model_path, "vocab.json"),
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eval=True,
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use_deepspeed=True
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)
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model.cuda()
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# This is for debugging purposes only
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DEVICE_ASSERT_DETECTED=0
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DEVICE_ASSERT_PROMPT=None
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DEVICE_ASSERT_LANG=None
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#supported_languages=["en","es","fr","de","it","pt","pl","tr","ru","nl","cs","ar","zh-cn"]
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supported_languages=config.languages
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def predict(
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if agree == True:
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-
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if language not in supported_languages:
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gr.Warning(
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return (
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-
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-
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-
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-
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language_predicted=langid.classify(prompt)[
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# tts expects chinese as zh-cn
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if language_predicted == "zh":
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#we use zh-cn
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language_predicted = "zh-cn"
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print(f"Detected language:{language_predicted}, Chosen language:{language}")
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# After text character length 15 trigger language detection
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if len(prompt)>15:
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# allow any language for short text as some may be common
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# If user unchecks language autodetection it will not trigger
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# You may remove this completely for own use
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if language_predicted != language and not no_lang_auto_detect:
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#Please duplicate and remove this check if you really want this
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#Or auto-detector fails to identify language (which it can on pretty short text or mixed text)
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gr.Warning(
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return (
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-
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-
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-
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-
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-
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-
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if use_mic == True:
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if mic_file_path is not None:
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-
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else:
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gr.Warning(
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return (
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None,
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None,
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None,
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None,
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)
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-
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else:
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speaker_wav=audio_file_pth
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-
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# Filtering for microphone input, as it has BG noise, maybe silence in beginning and end
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# This is fast filtering not perfect
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# Apply all on demand
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lowpassfilter=denoise=trim=loudness=True
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if lowpassfilter:
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lowpass_highpass="lowpass=8000,highpass=75,"
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else:
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lowpass_highpass=""
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if trim:
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# better to remove silence in beginning and end for microphone
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trim_silence="areverse,silenceremove=start_periods=1:start_silence=0:start_threshold=0.02,areverse,silenceremove=start_periods=1:start_silence=0:start_threshold=0.02,"
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else:
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trim_silence=""
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-
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if
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try:
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out_filename =
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#
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-
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print("Filtered microphone input")
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except subprocess.CalledProcessError:
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# There was an error - command exited with non-zero code
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print("Error: failed filtering, use original microphone input")
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else:
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speaker_wav=speaker_wav
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if len(prompt)<2:
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gr.Warning("Please give a longer prompt text")
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return (
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-
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if len(prompt)>200:
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gr.Warning(
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return (
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-
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global DEVICE_ASSERT_DETECTED
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if DEVICE_ASSERT_DETECTED:
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global DEVICE_ASSERT_PROMPT
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global DEVICE_ASSERT_LANG
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#It will likely never come here as we restart space on first unrecoverable error now
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print(
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# note diffusion_conditioning not used on hifigan (default mode), it will be empty but need to pass it to model.inference
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try:
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except Exception as e:
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print("Speaker encoding error", str(e))
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gr.Warning(
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return (
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None,
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None,
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None,
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None,
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)
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latent_calculation_time = time.time() - t_latent
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#metrics_text=f"Embedding calculation time: {latent_calculation_time:.2f} seconds\n"
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wav_chunks = []
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-
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print("I: Generating new audio...")
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t0 = time.time()
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out = model.inference(
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@@ -230,29 +264,78 @@ def predict(prompt, language, audio_file_pth, mic_file_path, use_mic, voice_clea
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print(f"Real-time factor (RTF): {real_time_factor}")
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metrics_text+=f"Real-time factor (RTF): {real_time_factor:.2f}\n"
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torchaudio.save("output.wav", torch.tensor(out["wav"]).unsqueeze(0), 24000)
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-
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if "device-side assert" in str(e):
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# cannot do anything on cuda device side error, need tor estart
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print(
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gr.Warning("Unhandled Exception encounter, please retry in a minute")
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print("Cuda device-assert Runtime encountered need restart")
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if not DEVICE_ASSERT_DETECTED:
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DEVICE_ASSERT_DETECTED=1
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DEVICE_ASSERT_PROMPT=prompt
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DEVICE_ASSERT_LANG=language
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-
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# just before restarting save what caused the issue so we can handle it in future
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# Uploading Error data only happens for unrecovarable error
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error_time = datetime.datetime.now().strftime(
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error_data = [
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print(error_data)
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print(speaker_wav)
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write_io = StringIO()
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csv.writer(write_io).writerows([error_data])
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csv_upload= write_io.getvalue().encode()
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filename =
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print("Writing error csv")
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error_api = HfApi()
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error_api.upload_file(
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@@ -261,10 +344,12 @@ def predict(prompt, language, audio_file_pth, mic_file_path, use_mic, voice_clea
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repo_id="coqui/xtts-flagged-dataset",
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repo_type="dataset",
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)
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-
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#speaker_wav
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print("Writing error reference audio")
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speaker_filename =
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error_api = HfApi()
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error_api.upload_file(
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path_or_fileobj=speaker_wav,
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@@ -273,21 +358,23 @@ def predict(prompt, language, audio_file_pth, mic_file_path, use_mic, voice_clea
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repo_type="dataset",
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)
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# HF Space specific.. This error is unrecoverable need to restart space
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api.restart_space(repo_id=repo_id)
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else:
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if "Failed to decode" in str(e):
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print("Speaker encoding error", str(e))
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gr.Warning(
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else:
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print("RuntimeError: non device-side assert error:", str(e))
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gr.Warning("Something unexpected happened please retry again.")
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return (
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-
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return (
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gr.make_waveform(
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audio="output.wav",
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@@ -299,11 +386,11 @@ def predict(prompt, language, audio_file_pth, mic_file_path, use_mic, voice_clea
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else:
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gr.Warning("Please accept the Terms & Condition!")
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return (
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-
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title = "Coqui🐸 XTTS"
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@@ -351,7 +438,6 @@ examples = [
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False,
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False,
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True,
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-
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],
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[
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"Lorsque j'avais six ans j'ai vu, une fois, une magnifique image",
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"it",
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"examples/female.wav",
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None,
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-
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False,
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False,
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True,
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@@ -428,7 +514,7 @@ examples = [
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"ru",
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"examples/female.wav",
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None,
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-
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False,
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False,
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True,
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@@ -438,7 +524,7 @@ examples = [
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"nl",
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"examples/male.wav",
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None,
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-
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False,
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False,
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True,
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"cs",
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"examples/female.wav",
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None,
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-
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False,
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False,
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True,
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@@ -458,7 +544,7 @@ examples = [
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"zh-cn",
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"examples/female.wav",
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None,
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-
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False,
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False,
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True,
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@@ -476,7 +562,6 @@ examples = [
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]
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-
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gr.Interface(
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fn=predict,
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inputs=[
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@@ -502,7 +587,7 @@ gr.Interface(
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"cs",
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"ar",
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"zh-cn",
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"ja"
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],
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max_choices=1,
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value="en",
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type="filepath",
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value="examples/female.wav",
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),
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gr.Audio(
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-
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gr.Checkbox(
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gr.Checkbox(
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label="Agree",
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value=False,
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info="I agree to the terms of the Coqui Public Model License at https://coqui.ai/cpml",
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),
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-
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-
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],
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outputs=[
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gr.Video(label="Waveform Visual"),
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gr.Audio(label="Synthesised Audio",autoplay=True),
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gr.Text(label="Metrics"),
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gr.Audio(label="Reference Audio Used"),
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],
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@@ -545,4 +635,5 @@ gr.Interface(
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description=description,
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article=article,
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examples=examples,
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).queue().launch(debug=True,show_api=True)
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import subprocess
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import random
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from zipfile import ZipFile
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+
import uuid
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import time
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import torch
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import torchaudio
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+
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# By using XTTS you agree to CPML license https://coqui.ai/cpml
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os.environ["COQUI_TOS_AGREED"] = "1"
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# langid is used to detect language for longer text
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# Most users expect text to be their own language, there is checkbox to disable it
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+
import langid
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import base64
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import csv
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from io import StringIO
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print("Export newer ffmpeg binary for denoise filter")
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ZipFile("ffmpeg.zip").extractall()
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print("Make ffmpeg binary executable")
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st = os.stat("ffmpeg")
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os.chmod("ffmpeg", st.st_mode | stat.S_IEXEC)
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# This will trigger downloading model
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print("Downloading if not downloaded Coqui XTTS V1.1")
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from TTS.utils.manage import ModelManager
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+
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model_name = "tts_models/multilingual/multi-dataset/xtts_v1.1"
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ModelManager().download_model(model_name)
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model_path = os.path.join(get_user_data_dir("tts"), model_name.replace("/", "--"))
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checkpoint_path=os.path.join(model_path, "model.pth"),
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vocab_path=os.path.join(model_path, "vocab.json"),
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eval=True,
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+
use_deepspeed=True,
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)
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model.cuda()
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# This is for debugging purposes only
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+
DEVICE_ASSERT_DETECTED = 0
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DEVICE_ASSERT_PROMPT = None
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DEVICE_ASSERT_LANG = None
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+
# supported_languages=["en","es","fr","de","it","pt","pl","tr","ru","nl","cs","ar","zh-cn"]
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supported_languages = config.languages
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def predict(
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prompt,
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language,
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audio_file_pth,
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mic_file_path,
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use_mic,
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voice_cleanup,
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no_lang_auto_detect,
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agree,
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):
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if agree == True:
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if language not in supported_languages:
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gr.Warning(
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f"Language you put {language} in is not in is not in our Supported Languages, please choose from dropdown"
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)
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+
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return (
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None,
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+
None,
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+
None,
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None,
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)
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language_predicted = langid.classify(prompt)[
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0
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].strip() # strip need as there is space at end!
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# tts expects chinese as zh-cn
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if language_predicted == "zh":
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# we use zh-cn
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language_predicted = "zh-cn"
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print(f"Detected language:{language_predicted}, Chosen language:{language}")
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# After text character length 15 trigger language detection
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117 |
+
if len(prompt) > 15:
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118 |
# allow any language for short text as some may be common
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119 |
# If user unchecks language autodetection it will not trigger
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120 |
# You may remove this completely for own use
|
121 |
if language_predicted != language and not no_lang_auto_detect:
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122 |
+
# Please duplicate and remove this check if you really want this
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123 |
+
# Or auto-detector fails to identify language (which it can on pretty short text or mixed text)
|
124 |
+
gr.Warning(
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125 |
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f"It looks like your text isn’t the language you chose , if you’re sure the text is the same language you chose, please check disable language auto-detection checkbox"
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)
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+
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return (
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None,
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+
None,
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+
None,
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+
None,
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+
)
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|
|
135 |
if use_mic == True:
|
136 |
if mic_file_path is not None:
|
137 |
+
speaker_wav = mic_file_path
|
138 |
else:
|
139 |
+
gr.Warning(
|
140 |
+
"Please record your voice with Microphone, or uncheck Use Microphone to use reference audios"
|
141 |
+
)
|
142 |
return (
|
143 |
None,
|
144 |
None,
|
145 |
None,
|
146 |
None,
|
147 |
+
)
|
148 |
+
|
149 |
else:
|
150 |
+
speaker_wav = audio_file_pth
|
151 |
|
|
|
152 |
# Filtering for microphone input, as it has BG noise, maybe silence in beginning and end
|
153 |
# This is fast filtering not perfect
|
154 |
|
155 |
# Apply all on demand
|
156 |
+
lowpassfilter = denoise = trim = loudness = True
|
157 |
+
|
158 |
if lowpassfilter:
|
159 |
+
lowpass_highpass = "lowpass=8000,highpass=75,"
|
160 |
else:
|
161 |
+
lowpass_highpass = ""
|
162 |
|
163 |
if trim:
|
164 |
# better to remove silence in beginning and end for microphone
|
165 |
+
trim_silence = "areverse,silenceremove=start_periods=1:start_silence=0:start_threshold=0.02,areverse,silenceremove=start_periods=1:start_silence=0:start_threshold=0.02,"
|
166 |
else:
|
167 |
+
trim_silence = ""
|
168 |
+
|
169 |
+
if voice_cleanup:
|
170 |
try:
|
171 |
+
out_filename = (
|
172 |
+
speaker_wav + str(uuid.uuid4()) + ".wav"
|
173 |
+
) # ffmpeg to know output format
|
174 |
+
|
175 |
+
# we will use newer ffmpeg as that has afftn denoise filter
|
176 |
+
shell_command = f"./ffmpeg -y -i {speaker_wav} -af {lowpass_highpass}{trim_silence} {out_filename}".split(
|
177 |
+
" "
|
178 |
+
)
|
179 |
+
|
180 |
+
command_result = subprocess.run(
|
181 |
+
[item for item in shell_command],
|
182 |
+
capture_output=False,
|
183 |
+
text=True,
|
184 |
+
check=True,
|
185 |
+
)
|
186 |
+
speaker_wav = out_filename
|
187 |
print("Filtered microphone input")
|
188 |
except subprocess.CalledProcessError:
|
189 |
# There was an error - command exited with non-zero code
|
190 |
print("Error: failed filtering, use original microphone input")
|
191 |
else:
|
192 |
+
speaker_wav = speaker_wav
|
193 |
|
194 |
+
if len(prompt) < 2:
|
195 |
gr.Warning("Please give a longer prompt text")
|
196 |
return (
|
197 |
+
None,
|
198 |
+
None,
|
199 |
+
None,
|
200 |
+
None,
|
201 |
+
)
|
202 |
+
if len(prompt) > 200:
|
203 |
+
gr.Warning(
|
204 |
+
"Text length limited to 200 characters for this demo, please try shorter text. You can clone this space and edit code for your own usage"
|
205 |
+
)
|
206 |
return (
|
207 |
+
None,
|
208 |
+
None,
|
209 |
+
None,
|
210 |
+
None,
|
211 |
+
)
|
212 |
global DEVICE_ASSERT_DETECTED
|
213 |
if DEVICE_ASSERT_DETECTED:
|
214 |
global DEVICE_ASSERT_PROMPT
|
215 |
global DEVICE_ASSERT_LANG
|
216 |
+
# It will likely never come here as we restart space on first unrecoverable error now
|
217 |
+
print(
|
218 |
+
f"Unrecoverable exception caused by language:{DEVICE_ASSERT_LANG} prompt:{DEVICE_ASSERT_PROMPT}"
|
219 |
+
)
|
220 |
+
|
221 |
+
try:
|
222 |
+
metrics_text = ""
|
223 |
+
t_latent = time.time()
|
224 |
+
|
225 |
# note diffusion_conditioning not used on hifigan (default mode), it will be empty but need to pass it to model.inference
|
226 |
try:
|
227 |
+
(
|
228 |
+
gpt_cond_latent,
|
229 |
+
diffusion_conditioning,
|
230 |
+
speaker_embedding,
|
231 |
+
) = model.get_conditioning_latents(audio_path=speaker_wav)
|
232 |
except Exception as e:
|
233 |
print("Speaker encoding error", str(e))
|
234 |
+
gr.Warning(
|
235 |
+
"It appears something wrong with reference, did you unmute your microphone?"
|
236 |
+
)
|
237 |
return (
|
238 |
None,
|
239 |
None,
|
240 |
None,
|
241 |
None,
|
242 |
+
)
|
243 |
+
|
|
|
244 |
latent_calculation_time = time.time() - t_latent
|
245 |
+
# metrics_text=f"Embedding calculation time: {latent_calculation_time:.2f} seconds\n"
|
246 |
|
247 |
wav_chunks = []
|
248 |
+
## Direct mode
|
249 |
+
"""
|
250 |
print("I: Generating new audio...")
|
251 |
t0 = time.time()
|
252 |
out = model.inference(
|
|
|
264 |
print(f"Real-time factor (RTF): {real_time_factor}")
|
265 |
metrics_text+=f"Real-time factor (RTF): {real_time_factor:.2f}\n"
|
266 |
torchaudio.save("output.wav", torch.tensor(out["wav"]).unsqueeze(0), 24000)
|
267 |
+
"""
|
268 |
+
|
269 |
+
print("I: Generating new audio in streaming mode...")
|
270 |
+
t0 = time.time()
|
271 |
+
chunks = model.inference_stream(
|
272 |
+
prompt,
|
273 |
+
language,
|
274 |
+
gpt_cond_latent,
|
275 |
+
speaker_embedding,
|
276 |
+
decoder="ne_hifigan",
|
277 |
+
)
|
278 |
+
|
279 |
+
first_chunk = True
|
280 |
+
for i, chunk in enumerate(chunks):
|
281 |
+
if first_chunk:
|
282 |
+
first_chunk_time = time.time() - t0
|
283 |
+
metrics_text += f"Latency to first audio chunk: {round(first_chunk_time*1000)} milliseconds\n"
|
284 |
+
first_chunk = False
|
285 |
+
wav_chunks.append(chunk)
|
286 |
+
print(f"Received chunk {i} of audio length {chunk.shape[-1]}")
|
287 |
+
inference_time = time.time() - t0
|
288 |
+
print(
|
289 |
+
f"I: Time to generate audio: {round(inference_time*1000)} milliseconds"
|
290 |
+
)
|
291 |
+
metrics_text += (
|
292 |
+
f"Time to generate audio: {round(inference_time*1000)} milliseconds\n"
|
293 |
+
)
|
294 |
+
|
295 |
+
wav = torch.cat(wav_chunks, dim=0)
|
296 |
+
print(wav.shape)
|
297 |
+
real_time_factor = (time.time() - t0) / wav.shape[0] * 24000
|
298 |
+
print(f"Real-time factor (RTF): {real_time_factor}")
|
299 |
+
metrics_text += f"Real-time factor (RTF): {real_time_factor:.2f}\n"
|
300 |
+
|
301 |
+
torchaudio.save("output.wav", wav.squeeze().unsqueeze(0).cpu(), 24000)
|
302 |
+
|
303 |
+
except RuntimeError as e:
|
304 |
if "device-side assert" in str(e):
|
305 |
# cannot do anything on cuda device side error, need tor estart
|
306 |
+
print(
|
307 |
+
f"Exit due to: Unrecoverable exception caused by language:{language} prompt:{prompt}",
|
308 |
+
flush=True,
|
309 |
+
)
|
310 |
gr.Warning("Unhandled Exception encounter, please retry in a minute")
|
311 |
print("Cuda device-assert Runtime encountered need restart")
|
312 |
if not DEVICE_ASSERT_DETECTED:
|
313 |
+
DEVICE_ASSERT_DETECTED = 1
|
314 |
+
DEVICE_ASSERT_PROMPT = prompt
|
315 |
+
DEVICE_ASSERT_LANG = language
|
316 |
+
|
317 |
# just before restarting save what caused the issue so we can handle it in future
|
318 |
# Uploading Error data only happens for unrecovarable error
|
319 |
+
error_time = datetime.datetime.now().strftime("%d-%m-%Y-%H:%M:%S")
|
320 |
+
error_data = [
|
321 |
+
error_time,
|
322 |
+
prompt,
|
323 |
+
language,
|
324 |
+
audio_file_pth,
|
325 |
+
mic_file_path,
|
326 |
+
use_mic,
|
327 |
+
voice_cleanup,
|
328 |
+
no_lang_auto_detect,
|
329 |
+
agree,
|
330 |
+
]
|
331 |
+
error_data = [str(e) if type(e) != str else e for e in error_data]
|
332 |
print(error_data)
|
333 |
print(speaker_wav)
|
334 |
write_io = StringIO()
|
335 |
csv.writer(write_io).writerows([error_data])
|
336 |
+
csv_upload = write_io.getvalue().encode()
|
337 |
+
|
338 |
+
filename = error_time + "_" + str(uuid.uuid4()) + ".csv"
|
339 |
print("Writing error csv")
|
340 |
error_api = HfApi()
|
341 |
error_api.upload_file(
|
|
|
344 |
repo_id="coqui/xtts-flagged-dataset",
|
345 |
repo_type="dataset",
|
346 |
)
|
347 |
+
|
348 |
+
# speaker_wav
|
349 |
print("Writing error reference audio")
|
350 |
+
speaker_filename = (
|
351 |
+
error_time + "_reference_" + str(uuid.uuid4()) + ".wav"
|
352 |
+
)
|
353 |
error_api = HfApi()
|
354 |
error_api.upload_file(
|
355 |
path_or_fileobj=speaker_wav,
|
|
|
358 |
repo_type="dataset",
|
359 |
)
|
360 |
|
361 |
+
# HF Space specific.. This error is unrecoverable need to restart space
|
362 |
api.restart_space(repo_id=repo_id)
|
363 |
else:
|
364 |
if "Failed to decode" in str(e):
|
365 |
print("Speaker encoding error", str(e))
|
366 |
+
gr.Warning(
|
367 |
+
"It appears something wrong with reference, did you unmute your microphone?"
|
368 |
+
)
|
369 |
else:
|
370 |
print("RuntimeError: non device-side assert error:", str(e))
|
371 |
gr.Warning("Something unexpected happened please retry again.")
|
372 |
return (
|
373 |
+
None,
|
374 |
+
None,
|
375 |
+
None,
|
376 |
+
None,
|
377 |
+
)
|
378 |
return (
|
379 |
gr.make_waveform(
|
380 |
audio="output.wav",
|
|
|
386 |
else:
|
387 |
gr.Warning("Please accept the Terms & Condition!")
|
388 |
return (
|
389 |
+
None,
|
390 |
+
None,
|
391 |
+
None,
|
392 |
+
None,
|
393 |
+
)
|
394 |
|
395 |
|
396 |
title = "Coqui🐸 XTTS"
|
|
|
438 |
False,
|
439 |
False,
|
440 |
True,
|
|
|
441 |
],
|
442 |
[
|
443 |
"Lorsque j'avais six ans j'ai vu, une fois, une magnifique image",
|
|
|
494 |
"it",
|
495 |
"examples/female.wav",
|
496 |
None,
|
497 |
+
False,
|
498 |
False,
|
499 |
False,
|
500 |
True,
|
|
|
514 |
"ru",
|
515 |
"examples/female.wav",
|
516 |
None,
|
517 |
+
False,
|
518 |
False,
|
519 |
False,
|
520 |
True,
|
|
|
524 |
"nl",
|
525 |
"examples/male.wav",
|
526 |
None,
|
527 |
+
False,
|
528 |
False,
|
529 |
False,
|
530 |
True,
|
|
|
534 |
"cs",
|
535 |
"examples/female.wav",
|
536 |
None,
|
537 |
+
False,
|
538 |
False,
|
539 |
False,
|
540 |
True,
|
|
|
544 |
"zh-cn",
|
545 |
"examples/female.wav",
|
546 |
None,
|
547 |
+
False,
|
548 |
False,
|
549 |
False,
|
550 |
True,
|
|
|
562 |
]
|
563 |
|
564 |
|
|
|
565 |
gr.Interface(
|
566 |
fn=predict,
|
567 |
inputs=[
|
|
|
587 |
"cs",
|
588 |
"ar",
|
589 |
"zh-cn",
|
590 |
+
"ja",
|
591 |
],
|
592 |
max_choices=1,
|
593 |
value="en",
|
|
|
598 |
type="filepath",
|
599 |
value="examples/female.wav",
|
600 |
),
|
601 |
+
gr.Audio(
|
602 |
+
source="microphone",
|
603 |
+
type="filepath",
|
604 |
+
info="Use your microphone to record audio",
|
605 |
+
label="Use Microphone for Reference",
|
606 |
+
),
|
607 |
+
gr.Checkbox(
|
608 |
+
label="Use Microphone",
|
609 |
+
value=False,
|
610 |
+
info="Notice: Microphone input may not work properly under traffic",
|
611 |
+
),
|
612 |
+
gr.Checkbox(
|
613 |
+
label="Cleanup Reference Voice",
|
614 |
+
value=False,
|
615 |
+
info="This check can improve output if your microphone or reference voice is noisy",
|
616 |
+
),
|
617 |
+
gr.Checkbox(
|
618 |
+
label="Do not use language auto-detect",
|
619 |
+
value=False,
|
620 |
+
info="Check to disable language auto-detection",
|
621 |
+
),
|
622 |
gr.Checkbox(
|
623 |
label="Agree",
|
624 |
value=False,
|
625 |
info="I agree to the terms of the Coqui Public Model License at https://coqui.ai/cpml",
|
626 |
),
|
|
|
|
|
627 |
],
|
628 |
outputs=[
|
629 |
gr.Video(label="Waveform Visual"),
|
630 |
+
gr.Audio(label="Synthesised Audio", autoplay=True),
|
631 |
gr.Text(label="Metrics"),
|
632 |
gr.Audio(label="Reference Audio Used"),
|
633 |
],
|
|
|
635 |
description=description,
|
636 |
article=article,
|
637 |
examples=examples,
|
638 |
+
).queue().launch(debug=True, show_api=True)
|
639 |
+
|