Spaces:
Running
on
Zero
Running
on
Zero
add Global stop flag
Browse files
app.py
CHANGED
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@@ -5,31 +5,28 @@ import asyncio
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from fastrtc.webrtc import WebRTC
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from pydub import AudioSegment
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import time
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import
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import os # Added to check if file exists
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from gradio.utils import get_space
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import spaces
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from app.logger_config import logger as logging
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from app.utils import (
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generate_coturn_config
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)
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# --- Constants
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EXAMPLE_FILES = ["data/bonjour.wav", "data/bonjour2.wav"]
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streaming_should_stop = threading.Event()
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@spaces.GPU
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def read_and_stream_audio(filepath_to_stream: str):
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"""
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-
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and streams it in 1-second chunks.
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"""
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-
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if not filepath_to_stream or not os.path.exists(filepath_to_stream):
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logging.error(f"Audio file not found or not specified: {filepath_to_stream}")
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# Attempt to use the default file as a fallback
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if os.path.exists(DEFAULT_FILE):
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logging.warning(f"Using default file: {DEFAULT_FILE}")
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filepath_to_stream = DEFAULT_FILE
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@@ -39,92 +36,79 @@ def read_and_stream_audio(filepath_to_stream: str):
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logging.info(f"Preparing audio segment from: {filepath_to_stream}")
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streaming_should_stop.clear()
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-
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try:
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segment = AudioSegment.from_file(filepath_to_stream)
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chunk_duree_ms = 1000
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logging.info(f"Starting streaming in {chunk_duree_ms}ms chunks...")
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for i, chunk in enumerate(segment[::chunk_duree_ms]):
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iter_start_time = time.perf_counter()
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logging.info(f"Sending chunk {i+1}...")
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-
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if streaming_should_stop.is_set():
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logging.info("Stop
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break
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output_chunk = (
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chunk.frame_rate,
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np.array(chunk.get_array_of_samples()).reshape(1, -1),
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)
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-
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yield output_chunk
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-
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-
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-
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sleep_duration = (chunk_duree_ms / 1000.0) - (processing_duration_ms / 1000.0) - 0.1
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if sleep_duration < 0:
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sleep_duration = 0.01 # Avoid negative sleep time
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logging.debug(f"Processing time: {processing_duration_ms:.2f}ms, Sleep: {sleep_duration:.2f}s")
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-
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# Using wait() allows the thread to wake up if the signal is received
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if streaming_should_stop.wait(timeout=sleep_duration):
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logging.info("Stop signal received while waiting.")
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break
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logging.info("Streaming finished.")
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except asyncio.CancelledError:
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logging.info("Stream
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raise
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except FileNotFoundError:
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logging.error(f"Critical error: File not found: {filepath_to_stream}")
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except Exception as e:
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logging.error(f"Error during
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raise
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finally:
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streaming_should_stop.clear()
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logging.info("Stop
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def stop_streaming():
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"""
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logging.info("Stop button clicked:
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streaming_should_stop.set()
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return None
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# --- Gradio Interface ---
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with gr.Blocks(theme=gr.themes.Soft()) as demo:
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gr.Markdown(
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"## Application 'Streamer' WebRTC (Serveur
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"
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"puis cliquez sur
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)
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# 1. State to store the path of the file to be read
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active_filepath = gr.State(value=DEFAULT_FILE)
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with gr.Row():
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with gr.Column():
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main_audio = gr.Audio(
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label="Source Audio",
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sources=["upload", "microphone"],
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type="filepath",
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value=DEFAULT_FILE,
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)
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with gr.Column():
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webrtc_stream = WebRTC(
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label="
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mode="receive",
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modality="audio",
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rtc_configuration=generate_coturn_config(),
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visible=True,
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height
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)
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-
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with gr.Row():
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with gr.Column():
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start_button = gr.Button("Start Streaming", variant="primary")
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@@ -133,38 +117,21 @@ with gr.Blocks(theme=gr.themes.Soft()) as demo:
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gr.Text()
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def set_new_file(filepath):
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"""Updates the state with the new path, or reverts to default if None."""
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if filepath is None:
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logging.info("Audio cleared, reverting to default example file.")
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-
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new_path = filepath
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# Returns the value to be put in the gr.State
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return new_path
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# Update the path if the user uploads, clears, or changes the file
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main_audio.change(
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fn=set_new_file,
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inputs=[main_audio],
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outputs=[active_filepath]
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)
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# Update the path if the user finishes a recording
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main_audio.stop_recording(
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fn=set_new_file,
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inputs=[main_audio],
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outputs=[active_filepath]
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)
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# Functions to update the interface state
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def start_streaming_ui():
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logging.info("UI: Starting stream. Disabling controls.")
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return {
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start_button: gr.Button(interactive=False),
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stop_button: gr.Button(interactive=True),
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main_audio: gr.Audio(visible=False),
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}
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def stop_streaming_ui():
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@@ -174,42 +141,38 @@ with gr.Blocks(theme=gr.themes.Soft()) as demo:
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stop_button: gr.Button(interactive=False),
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main_audio: gr.Audio(
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label="Source Audio",
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sources=["upload", "microphone"],
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type="filepath",
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value=active_filepath.value,
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visible=True
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ui_components = [
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start_button, stop_button,
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main_audio,
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]
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stream_event = webrtc_stream.stream(
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fn=read_and_stream_audio,
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inputs=[active_filepath],
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outputs=[webrtc_stream],
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trigger=start_button.click,
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concurrency_id="audio_stream",
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concurrency_limit=10
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)
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#
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start_button.click(
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fn=start_streaming_ui,
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outputs=ui_components
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)
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#
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stop_button.click(
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fn=stop_streaming,
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outputs=[webrtc_stream],
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).then(
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fn=stop_streaming_ui,
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outputs=ui_components
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)
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from fastrtc.webrtc import WebRTC
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from pydub import AudioSegment
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import time
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import os
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import spaces
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import threading
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from app.utils import generate_coturn_config
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# --- Constants ---
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EXAMPLE_FILES = ["data/bonjour.wav", "data/bonjour2.wav"]
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DEFAULT_FILE = EXAMPLE_FILES[0]
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# --- Global stop flag ---
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streaming_should_stop = threading.Event()
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# --- Audio Stream Function ---
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@spaces.GPU
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def read_and_stream_audio(filepath_to_stream: str):
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"""
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Stream an audio file in 1-second chunks until stop signal is received.
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"""
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if not filepath_to_stream or not os.path.exists(filepath_to_stream):
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logging.error(f"Audio file not found or not specified: {filepath_to_stream}")
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if os.path.exists(DEFAULT_FILE):
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logging.warning(f"Using default file: {DEFAULT_FILE}")
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filepath_to_stream = DEFAULT_FILE
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logging.info(f"Preparing audio segment from: {filepath_to_stream}")
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streaming_should_stop.clear()
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+
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try:
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segment = AudioSegment.from_file(filepath_to_stream)
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chunk_duree_ms = 1000
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logging.info(f"Starting streaming in {chunk_duree_ms}ms chunks...")
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for i, chunk in enumerate(segment[::chunk_duree_ms]):
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if streaming_should_stop.is_set():
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logging.info("Stop flag detected, ending stream.")
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break
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iter_start = time.perf_counter()
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logging.info(f"Sending chunk {i+1}...")
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+
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output_chunk = (
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chunk.frame_rate,
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np.array(chunk.get_array_of_samples()).reshape(1, -1),
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)
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yield output_chunk
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processing_duration_ms = (time.perf_counter() - iter_start) * 1000
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sleep_duration = max((chunk_duree_ms / 1000.0) - (processing_duration_ms / 1000.0) - 0.1, 0.01)
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time.sleep(sleep_duration)
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logging.info("Streaming finished successfully.")
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except asyncio.CancelledError:
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logging.info("Stream cancelled by user.")
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raise
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except Exception as e:
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logging.error(f"Error during audio streaming: {e}", exc_info=True)
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raise
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finally:
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streaming_should_stop.clear()
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logging.info("Stop flag cleared.")
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# --- Stop Function ---
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def stop_streaming():
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"""Set the stop flag to True."""
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logging.info("Stop button clicked: setting stop flag.")
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streaming_should_stop.set()
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return None
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# --- Gradio Interface ---
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with gr.Blocks(theme=gr.themes.Soft()) as demo:
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gr.Markdown(
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"## Application 'Streamer' WebRTC (Serveur → Client)\n"
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"Chargez un fichier, enregistrez depuis votre micro ou utilisez un exemple, "
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"puis cliquez sur **Start Streaming** pour écouter le flux en direct."
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)
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active_filepath = gr.State(value=DEFAULT_FILE)
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with gr.Row():
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with gr.Column():
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main_audio = gr.Audio(
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label="Source Audio",
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sources=["upload", "microphone"],
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type="filepath",
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value=DEFAULT_FILE,
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)
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with gr.Column():
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webrtc_stream = WebRTC(
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label="Flux Audio",
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mode="receive",
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modality="audio",
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rtc_configuration=generate_coturn_config(),
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visible=True,
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height=200,
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)
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with gr.Row():
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with gr.Column():
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start_button = gr.Button("Start Streaming", variant="primary")
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gr.Text()
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def set_new_file(filepath):
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if filepath is None:
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logging.info("Audio cleared, reverting to default example file.")
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return DEFAULT_FILE
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logging.info(f"New audio source selected: {filepath}")
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return filepath
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main_audio.change(fn=set_new_file, inputs=[main_audio], outputs=[active_filepath])
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main_audio.stop_recording(fn=set_new_file, inputs=[main_audio], outputs=[active_filepath])
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def start_streaming_ui():
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logging.info("UI: Starting stream. Disabling controls.")
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return {
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start_button: gr.Button(interactive=False),
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stop_button: gr.Button(interactive=True),
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main_audio: gr.Audio(visible=False),
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}
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def stop_streaming_ui():
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stop_button: gr.Button(interactive=False),
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main_audio: gr.Audio(
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label="Source Audio",
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sources=["upload", "microphone"],
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type="filepath",
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value=active_filepath.value,
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visible=True,
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),
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}
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ui_components = [start_button, stop_button, main_audio]
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# --- Stream event ---
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stream_event = webrtc_stream.stream(
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fn=read_and_stream_audio,
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inputs=[active_filepath],
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outputs=[webrtc_stream],
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trigger=start_button.click,
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concurrency_id="audio_stream",
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concurrency_limit=10,
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)
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# --- Button actions ---
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start_button.click(
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fn=start_streaming_ui,
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outputs=ui_components,
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)
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# ✅ Stop streaming instantly (thread-safe flag)
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stop_button.click(
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fn=stop_streaming,
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outputs=[webrtc_stream],
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).then(
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fn=stop_streaming_ui,
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outputs=ui_components,
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)
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