Multichannel streams

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import logging
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from typing import Type
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import threading
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import time
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import wave
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import numpy as np
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import pyaudio
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import assemblyai as aai
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from assemblyai.streaming.v3 import (
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    BeginEvent,
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    StreamingClient,
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    StreamingClientOptions,
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    StreamingError,
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    StreamingEvents,
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    StreamingParameters,
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    TerminationEvent,
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    TurnEvent,
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)
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# Configuration
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API_KEY = "<YOUR_API_KEY>"
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AUDIO_FILE_PATH = "<DUAL_CHANNEL_AUDIO_FILE_PATH>"
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logging.basicConfig(level=logging.INFO)
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logger = logging.getLogger(__name__)
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class ChannelTranscriber:
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    def __init__(self, channel_id, channel_name, sample_rate):
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        self.channel_id = channel_id
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        self.channel_name = channel_name
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        self.sample_rate = sample_rate
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        self.client = None
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        self.audio_data = []
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        self.current_turn_line = None
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        self.line_count = 0
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        self.streaming_done = threading.Event()
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    def load_audio_channel(self):
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        """Extract single channel from dual-channel audio file."""
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        with wave.open(AUDIO_FILE_PATH, 'rb') as wf:
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            frames = wf.readframes(wf.getnframes())
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            audio_array = np.frombuffer(frames, dtype=np.int16)
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            if wf.getnchannels() == 2:
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                audio_array = audio_array.reshape(-1, 2)
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                channel_audio = audio_array[:, self.channel_id]
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                # Split into chunks for streaming
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                FRAMES_PER_BUFFER = 400  # 50ms chunks
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                for i in range(0, len(channel_audio), FRAMES_PER_BUFFER):
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                    chunk = channel_audio[i:i+FRAMES_PER_BUFFER]
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                    if len(chunk) < FRAMES_PER_BUFFER:
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                        chunk = np.pad(chunk, (0, FRAMES_PER_BUFFER - len(chunk)), 'constant')
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                    self.audio_data.append(chunk.astype(np.int16).tobytes())
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    def clear_current_line(self):
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        if self.current_turn_line is not None:
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            print("\r" + " " * 100 + "\r", end="", flush=True)
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    def print_partial_transcript(self, words):
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        self.clear_current_line()
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        # Build transcript from individual words
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        word_texts = [word.text for word in words]
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        transcript = ' '.join(word_texts)
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        partial_text = f"{self.channel_name}: {transcript}"
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        print(partial_text, end="", flush=True)
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        self.current_turn_line = len(partial_text)
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    def print_final_transcript(self, transcript):
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        self.clear_current_line()
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        final_text = f"{self.channel_name}: {transcript}"
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        print(final_text, flush=True)
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        self.current_turn_line = None
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        self.line_count += 1
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    def on_begin(self, client: Type[StreamingClient], event: BeginEvent):
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        """Called when the streaming session begins."""
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        pass  # Session started
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    def on_turn(self, client: Type[StreamingClient], event: TurnEvent):
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        """Called when a turn is received."""
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        transcript = event.transcript.strip() if event.transcript else ''
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        formatted = event.turn_is_formatted
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        words = event.words if event.words else []
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        if transcript or words:
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            if formatted:
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                self.print_final_transcript(transcript)
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            else:
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                self.print_partial_transcript(words)
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    def on_terminated(self, client: Type[StreamingClient], event: TerminationEvent):
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        """Called when the session is terminated."""
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        self.clear_current_line()
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        self.streaming_done.set()
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    def on_error(self, client: Type[StreamingClient], error: StreamingError):
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        """Called when an error occurs."""
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        print(f"\n{self.channel_name}: Error: {error}")
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        self.streaming_done.set()
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    def start_transcription(self):
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        """Start the transcription for this channel."""
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        self.load_audio_channel()
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        # Create streaming client
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        self.client = StreamingClient(
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            StreamingClientOptions(
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                api_key=API_KEY,
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                api_host="streaming.assemblyai.com",
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            )
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        )
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        # Register event handlers
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        self.client.on(StreamingEvents.Begin, self.on_begin)
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        self.client.on(StreamingEvents.Turn, self.on_turn)
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        self.client.on(StreamingEvents.Termination, self.on_terminated)
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        self.client.on(StreamingEvents.Error, self.on_error)
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        # Connect to streaming service with turn detection configuration
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        self.client.connect(
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            StreamingParameters(
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                sample_rate=self.sample_rate,
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                format_turns=True,
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                end_of_turn_confidence_threshold=0.4,
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                min_end_of_turn_silence_when_confident=160,
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                max_turn_silence=400,
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            )
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        )
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        # Create audio generator
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        def audio_generator():
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            for chunk in self.audio_data:
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                yield chunk
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                time.sleep(0.05)  # 50ms intervals
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        try:
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            # Stream audio
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            self.client.stream(audio_generator())
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        finally:
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            # Disconnect
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            self.client.disconnect(terminate=True)
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            self.streaming_done.set()
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    def start_transcription_thread(self):
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        """Start transcription in a separate thread."""
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        thread = threading.Thread(target=self.start_transcription, daemon=True)
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        thread.start()
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        return thread
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def play_audio_file():
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    try:
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        with wave.open(AUDIO_FILE_PATH, 'rb') as wf:
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            p = pyaudio.PyAudio()
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            stream = p.open(
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                format=p.get_format_from_width(wf.getsampwidth()),
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                channels=wf.getnchannels(),
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                rate=wf.getframerate(),
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                output=True
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            )
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            print(f"Playing audio: {AUDIO_FILE_PATH}")
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            # Play audio in chunks
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            chunk_size = 1024
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            data = wf.readframes(chunk_size)
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            while data:
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                stream.write(data)
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                data = wf.readframes(chunk_size)
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            stream.stop_stream()
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            stream.close()
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            p.terminate()
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            print("Audio playback finished")
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    except Exception as e:
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        print(f"Error playing audio: {e}")
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def transcribe_multichannel():
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    # Get sample rate from file
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    with wave.open(AUDIO_FILE_PATH, 'rb') as wf:
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        sample_rate = wf.getframerate()
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    # Create transcribers for each channel
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    transcriber_1 = ChannelTranscriber(0, "Speaker 1", sample_rate)
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    transcriber_2 = ChannelTranscriber(1, "Speaker 2", sample_rate)
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    # Start audio playback
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    audio_thread = threading.Thread(target=play_audio_file, daemon=True)
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    audio_thread.start()
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    # Start both transcriptions
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    thread_1 = transcriber_1.start_transcription_thread()
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    thread_2 = transcriber_2.start_transcription_thread()
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    # Wait for completion
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    thread_1.join()
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    thread_2.join()
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    audio_thread.join()
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if __name__ == "__main__":
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    transcribe_multichannel()