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| # Copyright (c) Meta Platforms, Inc. and affiliates. | |
| # All rights reserved. | |
| # | |
| # This source code is licensed under the license found in the | |
| # LICENSE file in the root directory of this source tree. | |
| """Various utilities for audio convertion (pcm format, sample rate and channels), | |
| and volume normalization.""" | |
| import sys | |
| import typing as tp | |
| import julius | |
| import torch | |
| import torchaudio | |
| def convert_audio_channels(wav: torch.Tensor, channels: int = 2) -> torch.Tensor: | |
| """Convert audio to the given number of channels. | |
| Args: | |
| wav (torch.Tensor): Audio wave of shape [B, C, T]. | |
| channels (int): Expected number of channels as output. | |
| Returns: | |
| torch.Tensor: Downmixed or unchanged audio wave [B, C, T]. | |
| """ | |
| *shape, src_channels, length = wav.shape | |
| if src_channels == channels: | |
| pass | |
| elif channels == 1: | |
| # Case 1: | |
| # The caller asked 1-channel audio, and the stream has multiple | |
| # channels, downmix all channels. | |
| wav = wav.mean(dim=-2, keepdim=True) | |
| elif src_channels == 1: | |
| # Case 2: | |
| # The caller asked for multiple channels, but the input file has | |
| # a single channel, replicate the audio over all channels. | |
| wav = wav.expand(*shape, channels, length) | |
| elif src_channels >= channels: | |
| # Case 3: | |
| # The caller asked for multiple channels, and the input file has | |
| # more channels than requested. In that case return the first channels. | |
| wav = wav[..., :channels, :] | |
| else: | |
| # Case 4: What is a reasonable choice here? | |
| raise ValueError('The audio file has less channels than requested but is not mono.') | |
| return wav | |
| def convert_audio(wav: torch.Tensor, from_rate: float, | |
| to_rate: float, to_channels: int) -> torch.Tensor: | |
| """Convert audio to new sample rate and number of audio channels.""" | |
| wav = julius.resample_frac(wav, int(from_rate), int(to_rate)) | |
| wav = convert_audio_channels(wav, to_channels) | |
| return wav | |
| def normalize_loudness(wav: torch.Tensor, sample_rate: int, loudness_headroom_db: float = 14, | |
| loudness_compressor: bool = False, energy_floor: float = 2e-3): | |
| """Normalize an input signal to a user loudness in dB LKFS. | |
| Audio loudness is defined according to the ITU-R BS.1770-4 recommendation. | |
| Args: | |
| wav (torch.Tensor): Input multichannel audio data. | |
| sample_rate (int): Sample rate. | |
| loudness_headroom_db (float): Target loudness of the output in dB LUFS. | |
| loudness_compressor (bool): Uses tanh for soft clipping. | |
| energy_floor (float): anything below that RMS level will not be rescaled. | |
| Returns: | |
| torch.Tensor: Loudness normalized output data. | |
| """ | |
| energy = wav.pow(2).mean().sqrt().item() | |
| if energy < energy_floor: | |
| return wav | |
| transform = torchaudio.transforms.Loudness(sample_rate) | |
| input_loudness_db = transform(wav).item() | |
| # calculate the gain needed to scale to the desired loudness level | |
| delta_loudness = -loudness_headroom_db - input_loudness_db | |
| gain = 10.0 ** (delta_loudness / 20.0) | |
| output = gain * wav | |
| if loudness_compressor: | |
| output = torch.tanh(output) | |
| assert output.isfinite().all(), (input_loudness_db, wav.pow(2).mean().sqrt()) | |
| return output | |
| def _clip_wav(wav: torch.Tensor, log_clipping: bool = False, stem_name: tp.Optional[str] = None) -> None: | |
| """Utility function to clip the audio with logging if specified.""" | |
| max_scale = wav.abs().max() | |
| if log_clipping and max_scale > 1: | |
| clamp_prob = (wav.abs() > 1).float().mean().item() | |
| print(f"CLIPPING {stem_name or ''} happening with proba (a bit of clipping is okay):", | |
| clamp_prob, "maximum scale: ", max_scale.item(), file=sys.stderr) | |
| wav.clamp_(-1, 1) | |
| def normalize_audio(wav: torch.Tensor, normalize: bool = True, | |
| strategy: str = 'peak', peak_clip_headroom_db: float = 1, | |
| rms_headroom_db: float = 18, loudness_headroom_db: float = 14, | |
| loudness_compressor: bool = False, log_clipping: bool = False, | |
| sample_rate: tp.Optional[int] = None, | |
| stem_name: tp.Optional[str] = None) -> torch.Tensor: | |
| """Normalize the audio according to the prescribed strategy (see after). | |
| Args: | |
| wav (torch.Tensor): Audio data. | |
| normalize (bool): if `True` (default), normalizes according to the prescribed | |
| strategy (see after). If `False`, the strategy is only used in case clipping | |
| would happen. | |
| strategy (str): Can be either 'clip', 'peak', or 'rms'. Default is 'peak', | |
| i.e. audio is normalized by its largest value. RMS normalizes by root-mean-square | |
| with extra headroom to avoid clipping. 'clip' just clips. | |
| peak_clip_headroom_db (float): Headroom in dB when doing 'peak' or 'clip' strategy. | |
| rms_headroom_db (float): Headroom in dB when doing 'rms' strategy. This must be much larger | |
| than the `peak_clip` one to avoid further clipping. | |
| loudness_headroom_db (float): Target loudness for loudness normalization. | |
| loudness_compressor (bool): If True, uses tanh based soft clipping. | |
| log_clipping (bool): If True, basic logging on stderr when clipping still | |
| occurs despite strategy (only for 'rms'). | |
| sample_rate (int): Sample rate for the audio data (required for loudness). | |
| stem_name (str, optional): Stem name for clipping logging. | |
| Returns: | |
| torch.Tensor: Normalized audio. | |
| """ | |
| scale_peak = 10 ** (-peak_clip_headroom_db / 20) | |
| scale_rms = 10 ** (-rms_headroom_db / 20) | |
| if strategy == 'peak': | |
| rescaling = (scale_peak / wav.abs().max()) | |
| if normalize or rescaling < 1: | |
| wav = wav * rescaling | |
| elif strategy == 'clip': | |
| wav = wav.clamp(-scale_peak, scale_peak) | |
| elif strategy == 'rms': | |
| mono = wav.mean(dim=0) | |
| rescaling = scale_rms / mono.pow(2).mean().sqrt() | |
| if normalize or rescaling < 1: | |
| wav = wav * rescaling | |
| _clip_wav(wav, log_clipping=log_clipping, stem_name=stem_name) | |
| elif strategy == 'loudness': | |
| assert sample_rate is not None, "Loudness normalization requires sample rate." | |
| wav = normalize_loudness(wav, sample_rate, loudness_headroom_db, loudness_compressor) | |
| _clip_wav(wav, log_clipping=log_clipping, stem_name=stem_name) | |
| else: | |
| assert wav.abs().max() < 1 | |
| assert strategy == '' or strategy == 'none', f"Unexpected strategy: '{strategy}'" | |
| return wav | |
| def f32_pcm(wav: torch.Tensor) -> torch.Tensor: | |
| """Convert audio to float 32 bits PCM format. | |
| """ | |
| if wav.dtype.is_floating_point: | |
| return wav | |
| elif wav.dtype == torch.int16: | |
| return wav.float() / 2**15 | |
| elif wav.dtype == torch.int32: | |
| return wav.float() / 2**31 | |
| raise ValueError(f"Unsupported wav dtype: {wav.dtype}") | |
| def i16_pcm(wav: torch.Tensor) -> torch.Tensor: | |
| """Convert audio to int 16 bits PCM format. | |
| ..Warning:: There exist many formula for doing this conversion. None are perfect | |
| due to the asymmetry of the int16 range. One either have possible clipping, DC offset, | |
| or inconsistencies with f32_pcm. If the given wav doesn't have enough headroom, | |
| it is possible that `i16_pcm(f32_pcm)) != Identity`. | |
| """ | |
| if wav.dtype.is_floating_point: | |
| assert wav.abs().max() <= 1 | |
| candidate = (wav * 2 ** 15).round() | |
| if candidate.max() >= 2 ** 15: # clipping would occur | |
| candidate = (wav * (2 ** 15 - 1)).round() | |
| return candidate.short() | |
| else: | |
| assert wav.dtype == torch.int16 | |
| return wav | |