"""메모리 위에서만 도는 오디오 처리. 파일도 서브프로세스도 없다."""
import io

import av
import numpy as np

SAMPLE_RATE = 44100                    # 렌더 믹스 기준
MP3_BIT_RATE = 192000
INT16_FULL_SCALE = 32768


def decode_mp3(data: bytes, sample_rate: int = SAMPLE_RATE) -> np.ndarray:
    """mp3 → 모노 s16 PCM. 소스 샘플레이트가 달라도 리샘플해 맞춘다."""
    container = av.open(io.BytesIO(data))
    resampler = av.audio.resampler.AudioResampler(
        format="s16", layout="mono", rate=sample_rate)
    chunks = []
    for frame in container.decode(audio=0):
        for resampled in resampler.resample(frame):
            chunks.append(resampled.to_ndarray().reshape(-1))
    for resampled in resampler.resample(None):        # 남은 것 밀어내기
        chunks.append(resampled.to_ndarray().reshape(-1))
    container.close()
    return np.concatenate(chunks) if chunks else np.zeros(0, np.int16)


def encode_mp3(pcm: np.ndarray, sample_rate: int = SAMPLE_RATE) -> bytes:
    buffer = io.BytesIO()
    container = av.open(buffer, "w", format="mp3")
    stream = container.add_stream("mp3", rate=sample_rate)
    stream.layout = "mono"
    stream.bit_rate = MP3_BIT_RATE
    frame = av.AudioFrame.from_ndarray(pcm.reshape(1, -1), format="s16", layout="mono")
    frame.rate = sample_rate
    for packet in stream.encode(frame):
        container.mux(packet)
    for packet in stream.encode():
        container.mux(packet)
    container.close()
    return buffer.getvalue()


def trim_silence(pcm: np.ndarray, threshold_db: float = -45,
                 keep_lead: float = 0.05, keep_trail: float = 0.12,
                 sample_rate: int = SAMPLE_RATE) -> np.ndarray:
    """앞뒤 무음만 깎는다. 말소리 사이의 쉼은 보존한다."""
    threshold = INT16_FULL_SCALE * 10 ** (threshold_db / 20)
    loud = np.flatnonzero(np.abs(pcm) > threshold)
    if len(loud) == 0:
        return pcm[:0]
    start = max(0, loud[0] - round(keep_lead * sample_rate))
    end = min(len(pcm), loud[-1] + 1 + round(keep_trail * sample_rate))
    return pcm[start:end]