"""MP3 encode via PyAV/libmp3lame. ffmpeg CLI is not installed on this cluster; PyAV 18 bundles libmp3lame, so this is the only available CBR MP3 path. 160 kbps CBR, mono, 48 kHz.""" import io, numpy as np try: import av _HAVE_AV = True except ImportError: # env_asr (NeMo) has no PyAV; libsndfile 1.2 reads mp3 fine _HAVE_AV = False def encode_mp3(wav, sr=48000, bitrate=160000): """wav: float32 mono numpy in [-1,1] -> bytes of a 160 kbps CBR mono mp3.""" if not _HAVE_AV: raise RuntimeError("MP3 encoding needs PyAV (libmp3lame); this env has none") buf = io.BytesIO() c = av.open(buf, "w", format="mp3") st = c.add_stream("libmp3lame", rate=sr) st.bit_rate = bitrate st.bit_rate_tolerance = 0 # CBR, not ABR/VBR st.codec_context.layout = "mono" x = np.clip(np.asarray(wav, dtype=np.float32), -1.0, 1.0) x = (x * 32767.0).astype(np.int16).reshape(1, -1) fr = av.AudioFrame.from_ndarray(x, format="s16", layout="mono") fr.sample_rate = sr; fr.pts = 0 for p in st.encode(fr): c.mux(p) for p in st.encode(None): c.mux(p) c.close() return buf.getvalue() def decode_mp3(b): if not _HAVE_AV: import soundfile as sf x, sr = sf.read(io.BytesIO(b), dtype="float32") if x.ndim > 1: x = x.mean(1) return x.astype(np.float32), sr c = av.open(io.BytesIO(b)) st = c.streams.audio[0] nch = st.channels or 1 # STEREO BUG (fixed 2026-08-21). The old body was # fr = [f.to_ndarray().reshape(-1) for f in c.decode(audio=0)] # which is only correct for MONO. PyAV hands back either (channels, nb_samples) for planar # formats or (1, nb_samples*channels) for packed ones; a blind reshape(-1) turns BOTH into a # signal of length nb_samples*channels instead of nb_samples. The voice-profile audio is # duplicated mono written as 2-channel mp3, so every sample appeared twice and the decoded # signal was exactly 2x too long -- i.e. HALF SPEED. dur_s and moss_frames came out 2x on # every vprof_* row, and because BOTH were doubled the internal consistency check # `moss_frames == floor(dur_s * 12.5)` still passed, which is why this hid for so long. # The MOSS codes for those rows encode half-speed audio and are unusable; they must be # re-encoded. The soundfile fallback above was always correct (it does `x.mean(1)`). fr = [] for f in c.decode(audio=0): a = f.to_ndarray() if nch > 1: if a.ndim == 2 and a.shape[0] == nch: # planar: (channels, nb_samples) a = a.mean(0) else: # packed: (1, nb_samples*channels) a = a.reshape(-1, nch).mean(1) else: a = a.reshape(-1) fr.append(a) sr = st.rate; c.close() x = np.concatenate(fr) if fr else np.zeros(0, np.float32) if x.dtype == np.int16: x = x.astype(np.float32)/32768.0 return x.astype(np.float32), sr