Datasets:
File size: 3,005 Bytes
c413e6f | 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 | """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
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