"""Audio conversion service""" import struct from io import BytesIO import numpy as np import scipy.io.wavfile as wavfile import soundfile as sf from loguru import logger from pydub import AudioSegment from ..core.config import settings from .streaming_audio_writer import StreamingAudioWriter class AudioNormalizer: """Handles audio normalization state for a single stream""" def __init__(self): self.chunk_trim_ms = settings.gap_trim_ms self.sample_rate = 24000 # Sample rate of the audio self.samples_to_trim = int(self.chunk_trim_ms * self.sample_rate / 1000) async def normalize(self, audio_data: np.ndarray) -> np.ndarray: """Convert audio data to int16 range and trim silence from start and end Args: audio_data: Input audio data as numpy array Returns: Normalized and trimmed audio data """ if len(audio_data) == 0: raise ValueError("Empty audio data") # Trim start and end if enough samples if len(audio_data) > (2 * self.samples_to_trim): audio_data = audio_data[self.samples_to_trim : -self.samples_to_trim] # Scale directly to int16 range with clipping return np.clip(audio_data * 32767, -32768, 32767).astype(np.int16) class AudioService: """Service for audio format conversions with streaming support""" # Supported formats SUPPORTED_FORMATS = {"wav", "mp3", "opus", "flac", "aac", "pcm", "ogg"} # Default audio format settings balanced for speed and compression DEFAULT_SETTINGS = { "mp3": { "bitrate_mode": "CONSTANT", # Faster than variable bitrate "compression_level": 0.0, # Balanced compression }, "opus": { "compression_level": 0.0, # Good balance for speech }, "flac": { "compression_level": 0.0, # Light compression, still fast }, "aac": { "bitrate": "192k", # Default AAC bitrate }, } _writers = {} @staticmethod async def convert_audio( audio_data: np.ndarray, sample_rate: int, output_format: str, is_first_chunk: bool = True, is_last_chunk: bool = False, normalizer: AudioNormalizer = None, ) -> bytes: """Convert audio data to specified format with streaming support Args: audio_data: Numpy array of audio samples sample_rate: Sample rate of the audio output_format: Target format (wav, mp3, ogg, pcm) is_first_chunk: Whether this is the first chunk is_last_chunk: Whether this is the last chunk normalizer: Optional AudioNormalizer instance for consistent normalization Returns: Bytes of the converted audio chunk """ try: # Validate format if output_format not in AudioService.SUPPORTED_FORMATS: raise ValueError(f"Format {output_format} not supported") # Always normalize audio to ensure proper amplitude scaling if normalizer is None: normalizer = AudioNormalizer() normalized_audio = await normalizer.normalize(audio_data) # Get or create format-specific writer writer_key = f"{output_format}_{sample_rate}" if is_first_chunk or writer_key not in AudioService._writers: AudioService._writers[writer_key] = StreamingAudioWriter( output_format, sample_rate ) writer = AudioService._writers[writer_key] # Write audio data first if len(normalized_audio) > 0: chunk_data = writer.write_chunk(normalized_audio) # Then finalize if this is the last chunk if is_last_chunk: final_data = writer.write_chunk(finalize=True) del AudioService._writers[writer_key] return final_data if final_data else b"" return chunk_data if chunk_data else b"" except Exception as e: logger.error(f"Error converting audio stream to {output_format}: {str(e)}") raise ValueError( f"Failed to convert audio stream to {output_format}: {str(e)}" )