RealTimeSTT-fork/example_browserclient/server.py

if __name__ == '__main__':
    print("Starting server, please wait...")

    from RealtimeSTT import AudioToTextRecorder
    import asyncio
    import websockets
    import threading
    import numpy as np
    from scipy.signal import resample
    import json

    recorder = None
    recorder_ready = threading.Event()
    client_websocket = None

    async def send_to_client(message):
        if client_websocket:
            await client_websocket.send(message)

    def text_detected(text):
        asyncio.new_event_loop().run_until_complete(
            send_to_client(
                json.dumps({
                    'type': 'realtime',
                    'text': text
                })
            )
        )
        print(f"\r{text}", flush=True, end='')

    recorder_config = {
        'spinner': False,
        'use_microphone': False,
        'model': 'large-v2',
        'language': 'en',
        'silero_sensitivity': 0.4,
        'webrtc_sensitivity': 2,
        'post_speech_silence_duration': 0.7,
        'min_length_of_recording': 0,
        'min_gap_between_recordings': 0,
        'enable_realtime_transcription': True,
        'realtime_processing_pause': 0,
        'realtime_model_type': 'tiny.en',
        'on_realtime_transcription_stabilized': text_detected,
    }

    def recorder_thread():
        global recorder
        print("Initializing RealtimeSTT...")
        recorder = AudioToTextRecorder(**recorder_config)
        print("RealtimeSTT initialized")
        recorder_ready.set()
        while True:
            full_sentence = recorder.text()
            asyncio.new_event_loop().run_until_complete(
                send_to_client(
                    json.dumps({
                        'type': 'fullSentence',
                        'text': full_sentence
                    })
                )
            )
            print(f"\rSentence: {full_sentence}")

    def decode_and_resample(
            audio_data,
            original_sample_rate,
            target_sample_rate):

        # Decode 16-bit PCM data to numpy array
        audio_np = np.frombuffer(audio_data, dtype=np.int16)

        # Calculate the number of samples after resampling
        num_original_samples = len(audio_np)
        num_target_samples = int(num_original_samples * target_sample_rate /
                                 original_sample_rate)

        # Resample the audio
        resampled_audio = resample(audio_np, num_target_samples)

        return resampled_audio.astype(np.int16).tobytes()

    async def echo(websocket, path):
        print("Client connected")
        global client_websocket
        client_websocket = websocket
        async for message in websocket:

            if not recorder_ready.is_set():
                print("Recorder not ready")
                continue

            metadata_length = int.from_bytes(message[:4], byteorder='little')
            metadata_json = message[4:4+metadata_length].decode('utf-8')
            metadata = json.loads(metadata_json)
            sample_rate = metadata['sampleRate']
            chunk = message[4+metadata_length:]
            resampled_chunk = decode_and_resample(chunk, sample_rate, 16000)
            recorder.feed_audio(resampled_chunk)

    start_server = websockets.serve(echo, "0.0.0.0", 9001)

    recorder_thread = threading.Thread(target=recorder_thread)
    recorder_thread.start()
    recorder_ready.wait()

    print("Server started. Press Ctrl+C to stop the server.")
    asyncio.get_event_loop().run_until_complete(start_server)
    asyncio.get_event_loop().run_forever()