Commit eff7419f authored by Anika Treffehn's avatar Anika Treffehn
Browse files

Merge branch 'fix_length_output_16kHz_lp_function' into 'main' 

small fix in 16kHz lp filtering

See merge request !123
parents a2b69678 2f18a6b3

ivas_processing_scripts/audiotools/__init__.py

+4 −1
Original line number Diff line number Diff line
@@ -34,7 +34,10 @@ import argparse 
from itertools import repeat

from pathlib import Path



from ivas_processing_scripts.audiotools.constants import AUDIO_FORMATS, BINAURAL_LFE_GAIN

from ivas_processing_scripts.audiotools.constants import (

    AUDIO_FORMATS,

    BINAURAL_LFE_GAIN,

)

from ivas_processing_scripts.audiotools.convert import convert_file

from ivas_processing_scripts.utils import apply_func_parallel

other/lp16k.py

+10 −6
Original line number Diff line number Diff line
@@ -7,6 +7,7 @@ from time import sleep 


sys.path.append(str(Path(__file__).parent.parent))

from ivas_processing_scripts.audiotools.audio import fromfile

from ivas_processing_scripts.audiotools.audioarray import trim

from ivas_processing_scripts.audiotools.audiofile import write

from ivas_processing_scripts.audiotools.wrappers.filter import resample_itu

from ivas_processing_scripts.utils import progressbar_update, spinner
@@ -19,21 +20,24 @@ def lp16k(in_file, out_file): 
            f"Unsupported sampling rate {x.fs//1000} kHz for input file {in_file} - only 48 kHz is supported!"

        )



    # save length of audio since resampling can alter the length by one sample due to rounding

    len_signal = len(x.audio)



    # resample ITU only returns the audio array

    # but uses the sampling rate stored in the audio object

    # so we need to correctly set it after each call

    x.audio = resample_itu(x, 96000)

    x.fs = 96000



    x.audio = resample_itu(x, 32000)

    x.fs = 32000



    x.audio = resample_itu(x, 96000)

    x.fs = 96000



    x.audio = resample_itu(x, 48000)

    x.fs = 48000



    if len_signal != len(x.audio):

        # padding should usually not happen since the resampling only rounds up

        x.audio = trim(

            x.audio, x.fs, (0, len(x.audio) - len_signal), samples=True, pad_noise=True

        )



    write(out_file, x.audio, x.fs)