o i'@sddlmZddlZddlZddlZddlmZddlmZddl Z ddl m Z ddl m Z m Z ddlZddlZddlZddlZddlZddlZddlmZddlmZdd lmZddlZddlZddlZdfd d Zdfd dZdgddZ dhddZ!diddZ"de#de#fddZ$ddZ%dd Z&djd"d#Z'gd$ddggd%gdd&d'd&d(d(d)dfd*e e(d+e e(d,e)d-e e)d.e)d/e*d0e*d1e e)d2e e*d3e(d4e e(d5e)d6e+d7e*d8e+d9e+d:e+d;e e)e)fdZ,dkd?d@Z-dAdBZ.dldDdEZ/dFdGZ0ddHdIdIdIdIdIdJdKdLdMggfdNe e(dOe)dPe)d.e)dQe1dRe e(dSe e(fdTdUZ2dmdWdXZ3GdYdZdZZ4d[d\Z5d)d]e(d^e)d_e*d`e*fdadbZ6dndddeZ7dS)o) ThreadPoolN)partial)Path)tqdm)ListTuple) open_memmap read_audio) reverberate}c Csvt|}g}td|jd|D]%}||||}|jd|kr0t|t||jdf}||qt|S)a Takes an input list of 1D arrays (of different lengths), concatenates them together, and then extracts clips of a uniform size by dividing the combined array. 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Args: files (List[str]): A list of filepaths clip_size (int): The number of samples (of 16khz audio) for all of the rows in the array Returns: ndarray: A N by `clip_size` array with the audio data, converted to 16-bit PCM cSsg|]}|jdqSr)r.0rrrr ]z$load_audio_clips..Nr) rr ValueErrorsumr emptyrrsizeastypeint16) filesrrrNXZprevious_row_remainderZcntrowrrrload_audio_clipsCs*  r,ffmpegcCsD|dkrd|d|d}n |dkrd|d|d}t|dS)NZsoxzsox "z" -G -r 16000 -c 1 -b 16 ""r-zffmpeg -y -i "z " -ar 16000 ")ossystem)Z input_file output_filebackendcmdrrr _convert_clipvs  r4>r cCs6t|d}tt|d}||ddt||DdS)a9 Converts files in parallel with multithreading using Sox or ffmpeg. Intended to only convert input audio files into single-channel, 16 khz clips. Args: input_files (List[str]): A list of paths to input files output_files (List[str]): A list of paths to output files, corresponding 1:1 to the input files sr (int): The output sample rate of the converted clip ncpu (int): The number of CPUs to use for the conversion backend (str): The utilty to use for conversion, "sox" or "ffmpeg" Returns: None )Z processes)r2cSsg|]\}}||fqSrrrrjrrrrz!convert_clips..N)rrr4starmapzip)Z input_filesZ output_filessrZncpur2poolfrrr convert_clipss   r>r%c sg}g}|D]_}g}g} |r*ddt||D} || |dd| Dntt|D]} | | j|| j|| j q1|dkrU|t | |q|dkre|ddt| Dq|gkrfddt ||D} dd| Dd d| DfS|gfS) ah Gets the paths of wav files in flat target directories, automatically filtering out files below/above the specified length (in seconds). Assumes that all wav files are sampled at 16khz, are single channel, and have 16-bit PCM data. Uses `os.scandir` in Python for highly efficient file system exploration, and doesn't require loading the files into memory for length estimation. Args: target_dir (List[str]): The target directories containing the audio files min_length_secs (float): The minimum length in seconds (otherwise the clip is skipped) max_length_secs (float): The maximum length in seconds (otherwise the clip is skipped) duration_method (str): Whether to use the file size ('size'), or header information ('header') to estimate the duration of the audio file. 'size' is generally much faster, but assumes that all files in the target directory are the same type, sample rate, and bitrate. If None, durations are not calculated. glob_filter (str): A pathlib glob filter string to select specific files within the target directory Returns: tuple: A list of strings corresponding to the paths of the wav files that met the length criteria, and a list of their durations (in seconds) cSg|]}t|qSr)strrrrrrz&filter_audio_paths..cSsg|]}tj|qSr)r/pathgetsizerrrrrr8r%headercSr?r)get_clip_durationrrrrrrAcs(g|]\}}|kr|kr||fqSrrr6max_length_secsmin_length_secsrrrs(cSg|]}|dqSrrrrrrrrAcSrIr rrrrrrrA) rglobextendrr/scandirrrBstatst_sizeestimate_clip_durationr:) Z target_dirsrHrGZduration_methodZ glob_filter file_paths durations target_dirsizesZ dir_pathsrfilteredrrFrfilter_audio_pathss,   rV audio_filesrTcCslt|d}t|d}dtj|d|jj|jj}g}|D]}| |d||jjq$|S)aiEstimates the duration of each audio file in a list. Assumes that all of the audio files have the same audio format, bit depth, and sample rate. Args: audio_file (str): A list of audio file paths sizes (int): The size of each audio file in bytes Returns: list: A list of durations (in seconds) for the audio files r) torchaudioinfomutagenFiler/rBrCZbitratelengthr)rWrTdetailsZ correctionrRr%rrrrPs$rPcCsddd}d}zt|}Wn ty|YSwtj|}|jdkr2|jdkr0||d}|S|jdkrB|jdkrB||d }|S) a)Estimates the duration of an MP3 file from metadata and file-size. Is only accurate for 16000 khz sample rate audio with a relatively constant bit-rate. Args: fpath (str): The input path to the MP3 file Returns: float: The duration of the MP3 file in seconds g>&5?g>&%?)16_khz_single_channel 16_khz_stereorr r5r_r`)rYrZ RuntimeErrorr/rBrCZ num_channels sample_rate)ZfpathZconversion_factorsZduration_secondsmdnbytesrrrestimate_mp3_durations$        rfcCs0zt|}Wn tyYdSw|j|jS)zJGets the duration of an audio clip in seconds from file header informationr)rYrZrbZ num_framesrc)clipmetadatarrrrE s   rEracCs|d|dS)a Calculates the duration (in seconds) from a WAV file, assuming it contains 16 khz single-channel audio. The bit depth is user specified, and defaults to 2 for 16-bit PCM audio. Args: size (int): The file size in bytes nbytes (int): How many bytes for each data point in the audio (e.g., 16-bit is 2, 32-bit is 4, etc.) Returns: float: The duration of the audio file in seconds ,r5r)r%rerrrget_wav_duration_from_filesizes rj randomTgF)rrforeground_clipsbackground_clips combined_sizelabels batch_sizesnr_lowsnr_high start_indexforeground_durationsforeground_truncate_strategyrirsrir_probabilityvolume_augmentationgenerated_noise_augmentationshufflereturn_sequence_labelsreturn_background_clipsreturn_background_clips_delayseedc0 #s|rtj|t||sdg|}n t|dkr td|s)dgt|}|rYtjt|}t||}t||}t||}|rYt||}t dt||D]}d||||}dd||||D}dd|D}|rfddt |||||D}t||||}ddt ||D}d d|D}g}tj |d|d d }t |D]n\}}|jd||kr|t|||jdtj}|d|||<||d|||q|jd||kr5tj dtd |jd||}||||||<|||||||qtj|||}g} g}!t ||||D]Z\}"}#}$}%|#jd|kr\t|#jt|"|#|$|%}&|!t||%|%|"jdtj| krgd }'tjj|tj|'d }(t|(|(}(t|&|(tj|d}&| |&qIt| })tt|!}*| rtj| krt !t| \}+|+jdd kr|+t d|+jdd d d f}+t"|)|+dd})| rtjdd|)jd},|,|)jd ddd|)})ntjt#|)d dd\}-}.|)|-j$dd})|)%dtj&})tt'|)jd ddkd}/|)|/})||/}|*|/}*|sM|)|sG|n|*d fVqat|%dtj&|/}|)|sb|n|*|fVqad S)a Mixes foreground and background clips at a random SNR level in batches. References: https://pytorch.org/audio/main/tutorials/audio_data_augmentation_tutorial.html and https://speechbrain.readthedocs.io/en/latest/API/speechbrain.processing.speech_augmentation.html#speechbrain.processing.speech_augmentation.AddNoise Args: foreground_clips (List[str]): A list of paths to the foreground clips background_clips (List[str]): A list of paths to the background clips (randomly selected for each foreground clip) combined_size (int): The total length (in samples) of the combined clip. If needed, the background clips are duplicated or truncated to reach this length. labels (List[int]): A list of integer labels corresponding 1:1 for the foreground clips. Will be updated as needed with foreground clips to ensure that mixed clips retain the proper labels. batch_size (int): The batch size snr_low (float): The low SNR level of the mixing in db snr_high (float): The high snr level of the mixing in db start_index (List[int]): The starting position (in samples) for the foreground clip to start in the background clip. If the foreground clip is longer than `combined_size` when starting at this point, the foreground clip will be truncated according to the `foreground_truncate_strategy` argument. foreground_durations (List[float]): The desired duration of each foreground clip (in seconds) foreground_truncate_strategy (str): The method used to truncate the foreground clip, if needed based on the `start_index`, `foreground_durations`, and `combined_size` arguments. See the options in the `truncate_clip` method. rirs (List[str]): A list of paths to room impulse response functions (RIR) to convolve with the clips to simulate different recording environments. Applies a single random selection from the list RIR file to the entire batch. If empty (the default), nothing is done. rir_probability (float): The probability (between 0 and 1) that the batch will be convolved with a RIR file. volume_augmentation (bool): Whether to randomly apply volume augmentation to the clips in the batch. This simply scales the data of each clip such that the maximum value is is between 0.02 and 1.0 (the floor shouldn't be zero as beyond a certain point the audio data is no longer valid). generated_noise_augmentation: The probability of further mixing the mixed clip with generated random noise. Will be either "white", "brown", "blue", "pink", or "violet" noise, mixed at a random SNR between `snr_low` and `snr_high`. return_background_clips (bool): Whether to return the segment of the background clip that was mixed with each foreground clip in the batch. return_background_clips_delay (Tuple(int)): The lower and upper bound of a random delay (in samples) to apply to the segment of each returned backgroud clip mixed with each foreground clip in the batch. This is primarily intended to simulate the drift between input and output channels in audio devices, which means that the mixed audio is never exactly aligned with the two source clips. shuffle (bool): Whether to shuffle the foreground clips before mixing (default: True) return_sequence_labels (bool): Whether to return sequence labels (i.e., frame-level labels) for each clip based on the start/end positions of the foreground clip. seed (int): A random seed Returns: generator: Returns a generator that yields batches of mixed foreground/background audio, labels, and the background segments used for each audio clip (or None is the `return_backgroun_clips` argument is False) rzQError! At least one value of the `start_index` argument is <0. 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The individual probabilities of each augmentation may be adjusted with the "augmentation_probabilities" argument. Args: clip_paths (List[str]) = The input audio files (as paths) to augment. Note that these should be shorter than the "total_length" argument, else they will be truncated. total_length (int): The total length of audio files (in samples) after augmentation. All input clips will be left-padded with silence to reach this size, with between 0 and 200 ms of other audio after the end of the original input clip. sr (int): The sample size of the input audio files batch_size (int): The number of audio files to augment at once. augmentation_probabilities (dict): The individual probabilities of each augmentation. If all probabilities are zero, the input audio files will simply be padded with silence. THe default values are: { "SevenBandParametricEQ": 0.25, "TanhDistortion": 0.25, "PitchShift": 0.25, "BandStopFilter": 0.25, "AddColoredNoise": 0.25, "AddBackgroundNoise": 0.75, "Gain": 1.0, "RIR": 0.5 } background_clip_paths (List[str]) = The paths to background audio files to mix with the input files RIR_paths (List[str]) = The paths to room impulse response functions (RIRs) to convolve with the input files, producing a version of the input clip with different acoustic characteristics. Returns: ndarray: A batch of augmented audio clips of size (batch_size, total_length) ir)Z min_gain_dbZ max_gain_dbrg-C6?g?r)Zmin_distortionZmax_distortionrrr5Z per_batch)Zmin_transpose_semitonesZmax_transpose_semitonesrrcmoder)rrrrar) min_snr_in_db max_snr_in_dbZ min_f_decayZ max_f_decayrrri)rZbackground_pathsrrrrr)Zmax_gain_in_dbrz2Error! 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Returns: list: A list of strings corresponding to words and phrases that are phonetically similar (but not identical) to the input text. )ZAAZAEZAHZAOZAWZAXZAXRZAYZEHZERZEYZIHZIXZIYZOWZOYZUHZUWZUXcSsg|]}t|qSr pronouncingZphones_for_wordrrrrrrz.generate_adversarial_texts.. resourceszen_us_cmudict_forward.ptz;Downloading phonemizer model from DeepPhonemizer library...rNzchttps://public-asai-dl-models.s3.eu-central-1.amazonaws.com/DeepPhonemizer/en_us_cmudict_forward.ptT)streamwbi) chunk_size) Phonemizerz The word 'zj' was not found in the pronunciation dictionary! 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