do mixed-precision for AMP inside the compress function itself, because the loudness function gripes when using a float16 (non-power of 2 lengths) or bfloat16 (something about views for bfloat16)

vall_e/emb/qnt.py

@@ -36,6 +36,118 @@ try:
 	"""
 	from dac.model.base import CodecMixin
 
+	@torch.no_grad()
+	def CodecMixin_compress(
+		self,
+		audio_path_or_signal: Union[str, Path, AudioSignal],
+		win_duration: float = 1.0,
+		verbose: bool = False,
+		normalize_db: float = -16,
+		n_quantizers: int = None,
+	) -> DACFile:
+		"""Processes an audio signal from a file or AudioSignal object into
+		discrete codes. This function processes the signal in short windows,
+		using constant GPU memory.
+
+		Parameters
+		----------
+		audio_path_or_signal : Union[str, Path, AudioSignal]
+			audio signal to reconstruct
+		win_duration : float, optional
+			window duration in seconds, by default 5.0
+		verbose : bool, optional
+			by default False
+		normalize_db : float, optional
+			normalize db, by default -16
+
+		Returns
+		-------
+		DACFile
+			Object containing compressed codes and metadata
+			required for decompression
+		"""
+		audio_signal = audio_path_or_signal
+		if isinstance(audio_signal, (str, Path)):
+			audio_signal = AudioSignal.load_from_file_with_ffmpeg(str(audio_signal))
+
+		self.eval()
+		original_padding = self.padding
+		original_device = audio_signal.device
+
+		audio_signal = audio_signal.clone()
+		original_sr = audio_signal.sample_rate
+
+		resample_fn = audio_signal.resample
+		loudness_fn = audio_signal.loudness
+
+		# If audio is > 10 minutes long, use the ffmpeg versions
+		if audio_signal.signal_duration >= 10 * 60 * 60:
+			resample_fn = audio_signal.ffmpeg_resample
+			loudness_fn = audio_signal.ffmpeg_loudness
+
+		original_length = audio_signal.signal_length
+		resample_fn(self.sample_rate)
+		input_db = loudness_fn()
+
+		if normalize_db is not None:
+			audio_signal.normalize(normalize_db)
+		audio_signal.ensure_max_of_audio()
+
+		nb, nac, nt = audio_signal.audio_data.shape
+		audio_signal.audio_data = audio_signal.audio_data.reshape(nb * nac, 1, nt)
+		win_duration = (
+			audio_signal.signal_duration if win_duration is None else win_duration
+		)
+
+		if audio_signal.signal_duration <= win_duration:
+			# Unchunked compression (used if signal length < win duration)
+			self.padding = True
+			n_samples = nt
+			hop = nt
+		else:
+			# Chunked inference
+			self.padding = False
+			# Zero-pad signal on either side by the delay
+			audio_signal.zero_pad(self.delay, self.delay)
+			n_samples = int(win_duration * self.sample_rate)
+			# Round n_samples to nearest hop length multiple
+			n_samples = int(math.ceil(n_samples / self.hop_length) * self.hop_length)
+			hop = self.get_output_length(n_samples)
+
+		codes = []
+		range_fn = range if not verbose else tqdm.trange
+
+		for i in range_fn(0, nt, hop):
+			x = audio_signal[..., i : i + n_samples]
+			x = x.zero_pad(0, max(0, n_samples - x.shape[-1]))
+
+			audio_data = x.audio_data.to(self.device)
+			audio_data = self.preprocess(audio_data, self.sample_rate)
+			with torch.autocast("cuda", dtype=cfg.inference.dtype, enabled=cfg.inference.amp):
+				_, c, _, _, _ = self.encode(audio_data, n_quantizers)
+			codes.append(c.to(original_device))
+			chunk_length = c.shape[-1]
+
+		codes = torch.cat(codes, dim=-1)
+
+		dac_file = DACFile(
+			codes=codes,
+			chunk_length=chunk_length,
+			original_length=original_length,
+			input_db=input_db,
+			channels=nac,
+			sample_rate=original_sr,
+			padding=self.padding,
+			dac_version="1.0.0",
+			#dac_version=SUPPORTED_VERSIONS[-1],
+		)
+
+		if n_quantizers is not None:
+			codes = codes[:, :n_quantizers, :]
+
+		self.padding = original_padding
+		return dac_file
+
 	@torch.no_grad()
 	def CodecMixin_decompress(
 		self,
@@ -84,6 +196,7 @@ try:
 		self.padding = original_padding
 		return recons
 
+	CodecMixin.compress = CodecMixin_compress
 	CodecMixin.decompress = CodecMixin_decompress
 
 except Exception as e:
@@ -368,9 +481,8 @@ def encode(wav: Tensor, sr: int = cfg.sample_rate, device="cuda", levels=cfg.mod
 		if not isinstance(levels, int):
 			levels = 8 if model.model_type == "24khz" else None
 
-		with torch.autocast("cuda", dtype=cfg.inference.dtype, enabled=cfg.inference.amp):
+		artifact = model.compress(signal, win_duration=None, verbose=False, n_quantizers=levels)
-			artifact = model.compress(signal, win_duration=None, verbose=False, n_quantizers=levels)
+		#artifact = model.compress(signal, n_quantizers=levels)
-			#artifact = model.compress(signal, n_quantizers=levels)
 		return artifact.codes if not return_metadata else artifact
 
 	# AudioDec uses a different pathway