Spaces:
Sleeping
Sleeping
Michael Hu
commited on
Commit
Β·
2477bc4
1
Parent(s):
ee54430
update logging level
Browse files- app.py +7 -7
- utils/stt.py +5 -5
- utils/translation.py +3 -3
- utils/tts.py +3 -3
app.py
CHANGED
|
@@ -26,12 +26,12 @@ from utils.tts_dummy import generate_speech
|
|
| 26 |
# Hugging Face Spaces Setup Automation
|
| 27 |
def setup_huggingface_space():
|
| 28 |
"""Automatically configure Hugging Face Space requirements"""
|
| 29 |
-
logger.
|
| 30 |
st.sidebar.header("Space Configuration")
|
| 31 |
|
| 32 |
try:
|
| 33 |
subprocess.run(["espeak-ng", "--version"], check=True, capture_output=True)
|
| 34 |
-
logger.
|
| 35 |
except (FileNotFoundError, subprocess.CalledProcessError):
|
| 36 |
logger.error("Missing espeak-ng dependency")
|
| 37 |
st.sidebar.error("""
|
|
@@ -64,7 +64,7 @@ os.makedirs("temp/outputs", exist_ok=True)
|
|
| 64 |
|
| 65 |
def configure_page():
|
| 66 |
"""Set up Streamlit page configuration"""
|
| 67 |
-
logger.
|
| 68 |
st.set_page_config(
|
| 69 |
page_title="Audio Translator",
|
| 70 |
page_icon="π§",
|
|
@@ -93,7 +93,7 @@ def handle_file_processing(upload_path):
|
|
| 93 |
|
| 94 |
try:
|
| 95 |
# STT Phase
|
| 96 |
-
logger.
|
| 97 |
status_text.markdown("π **Performing Speech Recognition...**")
|
| 98 |
with st.spinner("Initializing Whisper model..."):
|
| 99 |
english_text = transcribe_audio(upload_path)
|
|
@@ -101,7 +101,7 @@ def handle_file_processing(upload_path):
|
|
| 101 |
logger.info(f"STT completed. Text length: {len(english_text)} characters")
|
| 102 |
|
| 103 |
# Translation Phase
|
| 104 |
-
logger.
|
| 105 |
status_text.markdown("π **Translating Content...**")
|
| 106 |
with st.spinner("Loading translation model..."):
|
| 107 |
chinese_text = translate_text(english_text)
|
|
@@ -109,7 +109,7 @@ def handle_file_processing(upload_path):
|
|
| 109 |
logger.info(f"Translation completed. Translated length: {len(chinese_text)} characters")
|
| 110 |
|
| 111 |
# TTS Phase
|
| 112 |
-
logger.
|
| 113 |
status_text.markdown("π΅ **Generating Chinese Speech...**")
|
| 114 |
with st.spinner("Initializing TTS engine..."):
|
| 115 |
output_path = generate_speech(chinese_text, language="zh")
|
|
@@ -131,7 +131,7 @@ def handle_file_processing(upload_path):
|
|
| 131 |
|
| 132 |
def render_results(english_text, chinese_text, output_path):
|
| 133 |
"""Display processing results in organized columns"""
|
| 134 |
-
logger.
|
| 135 |
st.divider()
|
| 136 |
|
| 137 |
col1, col2 = st.columns([2, 1])
|
|
|
|
| 26 |
# Hugging Face Spaces Setup Automation
|
| 27 |
def setup_huggingface_space():
|
| 28 |
"""Automatically configure Hugging Face Space requirements"""
|
| 29 |
+
logger.info("Running Hugging Face space setup")
|
| 30 |
st.sidebar.header("Space Configuration")
|
| 31 |
|
| 32 |
try:
|
| 33 |
subprocess.run(["espeak-ng", "--version"], check=True, capture_output=True)
|
| 34 |
+
logger.info("espeak-ng verification successful")
|
| 35 |
except (FileNotFoundError, subprocess.CalledProcessError):
|
| 36 |
logger.error("Missing espeak-ng dependency")
|
| 37 |
st.sidebar.error("""
|
|
|
|
| 64 |
|
| 65 |
def configure_page():
|
| 66 |
"""Set up Streamlit page configuration"""
|
| 67 |
+
logger.info("Configuring Streamlit page")
|
| 68 |
st.set_page_config(
|
| 69 |
page_title="Audio Translator",
|
| 70 |
page_icon="π§",
|
|
|
|
| 93 |
|
| 94 |
try:
|
| 95 |
# STT Phase
|
| 96 |
+
logger.info("Beginning STT processing")
|
| 97 |
status_text.markdown("π **Performing Speech Recognition...**")
|
| 98 |
with st.spinner("Initializing Whisper model..."):
|
| 99 |
english_text = transcribe_audio(upload_path)
|
|
|
|
| 101 |
logger.info(f"STT completed. Text length: {len(english_text)} characters")
|
| 102 |
|
| 103 |
# Translation Phase
|
| 104 |
+
logger.info("Beginning translation")
|
| 105 |
status_text.markdown("π **Translating Content...**")
|
| 106 |
with st.spinner("Loading translation model..."):
|
| 107 |
chinese_text = translate_text(english_text)
|
|
|
|
| 109 |
logger.info(f"Translation completed. Translated length: {len(chinese_text)} characters")
|
| 110 |
|
| 111 |
# TTS Phase
|
| 112 |
+
logger.info("Beginning TTS generation")
|
| 113 |
status_text.markdown("π΅ **Generating Chinese Speech...**")
|
| 114 |
with st.spinner("Initializing TTS engine..."):
|
| 115 |
output_path = generate_speech(chinese_text, language="zh")
|
|
|
|
| 131 |
|
| 132 |
def render_results(english_text, chinese_text, output_path):
|
| 133 |
"""Display processing results in organized columns"""
|
| 134 |
+
logger.info("Rendering results")
|
| 135 |
st.divider()
|
| 136 |
|
| 137 |
col1, col2 = st.columns([2, 1])
|
utils/stt.py
CHANGED
|
@@ -22,17 +22,17 @@ def transcribe_audio(audio_path):
|
|
| 22 |
|
| 23 |
try:
|
| 24 |
# Audio conversion
|
| 25 |
-
logger.
|
| 26 |
audio = AudioSegment.from_file(audio_path)
|
| 27 |
processed_audio = audio.set_frame_rate(16000).set_channels(1)
|
| 28 |
wav_path = audio_path.replace(".mp3", ".wav")
|
| 29 |
processed_audio.export(wav_path, format="wav")
|
| 30 |
-
logger.
|
| 31 |
|
| 32 |
# Model initialization
|
| 33 |
logger.info("Loading Whisper model")
|
| 34 |
device = "cuda" if torch.cuda.is_available() else "cpu"
|
| 35 |
-
logger.
|
| 36 |
|
| 37 |
model = AutoModelForSpeechSeq2Seq.from_pretrained(
|
| 38 |
"openai/whisper-large-v3",
|
|
@@ -42,10 +42,10 @@ def transcribe_audio(audio_path):
|
|
| 42 |
).to(device)
|
| 43 |
|
| 44 |
processor = AutoProcessor.from_pretrained("openai/whisper-large-v3")
|
| 45 |
-
logger.
|
| 46 |
|
| 47 |
# Processing
|
| 48 |
-
logger.
|
| 49 |
inputs = processor(
|
| 50 |
wav_path,
|
| 51 |
sampling_rate=16000,
|
|
|
|
| 22 |
|
| 23 |
try:
|
| 24 |
# Audio conversion
|
| 25 |
+
logger.info("Converting audio format")
|
| 26 |
audio = AudioSegment.from_file(audio_path)
|
| 27 |
processed_audio = audio.set_frame_rate(16000).set_channels(1)
|
| 28 |
wav_path = audio_path.replace(".mp3", ".wav")
|
| 29 |
processed_audio.export(wav_path, format="wav")
|
| 30 |
+
logger.info(f"Audio converted to: {wav_path}")
|
| 31 |
|
| 32 |
# Model initialization
|
| 33 |
logger.info("Loading Whisper model")
|
| 34 |
device = "cuda" if torch.cuda.is_available() else "cpu"
|
| 35 |
+
logger.info(f"Using device: {device}")
|
| 36 |
|
| 37 |
model = AutoModelForSpeechSeq2Seq.from_pretrained(
|
| 38 |
"openai/whisper-large-v3",
|
|
|
|
| 42 |
).to(device)
|
| 43 |
|
| 44 |
processor = AutoProcessor.from_pretrained("openai/whisper-large-v3")
|
| 45 |
+
logger.info("Model loaded successfully")
|
| 46 |
|
| 47 |
# Processing
|
| 48 |
+
logger.info("Processing audio input")
|
| 49 |
inputs = processor(
|
| 50 |
wav_path,
|
| 51 |
sampling_rate=16000,
|
utils/translation.py
CHANGED
|
@@ -23,7 +23,7 @@ def translate_text(text):
|
|
| 23 |
logger.info("Loading NLLB model")
|
| 24 |
tokenizer = AutoTokenizer.from_pretrained("facebook/nllb-200-3.3B")
|
| 25 |
model = AutoModelForSeq2SeqLM.from_pretrained("facebook/nllb-200-3.3B")
|
| 26 |
-
logger.
|
| 27 |
|
| 28 |
# Text processing
|
| 29 |
max_chunk_length = 1000
|
|
@@ -32,7 +32,7 @@ def translate_text(text):
|
|
| 32 |
|
| 33 |
translated_chunks = []
|
| 34 |
for i, chunk in enumerate(text_chunks):
|
| 35 |
-
logger.
|
| 36 |
inputs = tokenizer(
|
| 37 |
chunk,
|
| 38 |
return_tensors="pt",
|
|
@@ -47,7 +47,7 @@ def translate_text(text):
|
|
| 47 |
)
|
| 48 |
translated = tokenizer.decode(outputs[0], skip_special_tokens=True)
|
| 49 |
translated_chunks.append(translated)
|
| 50 |
-
logger.
|
| 51 |
|
| 52 |
result = "".join(translated_chunks)
|
| 53 |
logger.info(f"Translation completed. Total length: {len(result)}")
|
|
|
|
| 23 |
logger.info("Loading NLLB model")
|
| 24 |
tokenizer = AutoTokenizer.from_pretrained("facebook/nllb-200-3.3B")
|
| 25 |
model = AutoModelForSeq2SeqLM.from_pretrained("facebook/nllb-200-3.3B")
|
| 26 |
+
logger.info("Translation model loaded")
|
| 27 |
|
| 28 |
# Text processing
|
| 29 |
max_chunk_length = 1000
|
|
|
|
| 32 |
|
| 33 |
translated_chunks = []
|
| 34 |
for i, chunk in enumerate(text_chunks):
|
| 35 |
+
logger.info(f"Processing chunk {i+1}/{len(text_chunks)}")
|
| 36 |
inputs = tokenizer(
|
| 37 |
chunk,
|
| 38 |
return_tensors="pt",
|
|
|
|
| 47 |
)
|
| 48 |
translated = tokenizer.decode(outputs[0], skip_special_tokens=True)
|
| 49 |
translated_chunks.append(translated)
|
| 50 |
+
logger.info(f"Chunk {i+1} translated successfully")
|
| 51 |
|
| 52 |
result = "".join(translated_chunks)
|
| 53 |
logger.info(f"Translation completed. Total length: {len(result)}")
|
utils/tts.py
CHANGED
|
@@ -19,7 +19,7 @@ class TTSEngine:
|
|
| 19 |
def __init__(self):
|
| 20 |
logger.info("Initializing TTS Engine")
|
| 21 |
self.device = "cuda" if torch.cuda.is_available() else "cpu"
|
| 22 |
-
logger.
|
| 23 |
self._verify_model_files()
|
| 24 |
logger.info("Loading Kokoro model")
|
| 25 |
self.model = build_model(f"{MODEL_DIR}/kokoro-v0_19.pth", self.device)
|
|
@@ -56,7 +56,7 @@ class TTSEngine:
|
|
| 56 |
logger.warning(f"Truncating long text ({len(text)} characters)")
|
| 57 |
text = text[:495] + "[TRUNCATED]"
|
| 58 |
|
| 59 |
-
logger.
|
| 60 |
audio, _ = generate_full(
|
| 61 |
self.model,
|
| 62 |
text,
|
|
@@ -66,7 +66,7 @@ class TTSEngine:
|
|
| 66 |
)
|
| 67 |
|
| 68 |
output_path = f"temp/outputs/output_{int(time.time())}.wav"
|
| 69 |
-
logger.
|
| 70 |
AudioSegment(
|
| 71 |
audio.numpy().tobytes(),
|
| 72 |
frame_rate=24000,
|
|
|
|
| 19 |
def __init__(self):
|
| 20 |
logger.info("Initializing TTS Engine")
|
| 21 |
self.device = "cuda" if torch.cuda.is_available() else "cpu"
|
| 22 |
+
logger.info(f"Using device: {self.device}")
|
| 23 |
self._verify_model_files()
|
| 24 |
logger.info("Loading Kokoro model")
|
| 25 |
self.model = build_model(f"{MODEL_DIR}/kokoro-v0_19.pth", self.device)
|
|
|
|
| 56 |
logger.warning(f"Truncating long text ({len(text)} characters)")
|
| 57 |
text = text[:495] + "[TRUNCATED]"
|
| 58 |
|
| 59 |
+
logger.info("Starting audio generation")
|
| 60 |
audio, _ = generate_full(
|
| 61 |
self.model,
|
| 62 |
text,
|
|
|
|
| 66 |
)
|
| 67 |
|
| 68 |
output_path = f"temp/outputs/output_{int(time.time())}.wav"
|
| 69 |
+
logger.info(f"Saving audio to {output_path}")
|
| 70 |
AudioSegment(
|
| 71 |
audio.numpy().tobytes(),
|
| 72 |
frame_rate=24000,
|