FreeDATA/modem/data cemetery/test/test_resample_48_8.py
2023-12-17 12:35:21 +01:00

120 lines
3.8 KiB
Python

#!/usr/bin/env python3
# -*- coding: utf-8 -*-
#
# Unit test for FreeDV API resampler functions, from
# codec2/unittest/t48_8_short.c - generate a sine wave at 8 KHz,
# upsample to 48 kHz, add an interferer, then downsample back to 8 kHz
#
# You can listen to the output files with:
#
# aplay -f S16_LE in8.raw
# aplay -r 48000 -f S16_LE out48.raw
# aplay -f S16_LE out8.raw
#
# They should sound like clean sine waves
# pylint: disable=global-statement, invalid-name, unused-import
import os
import sys
import codec2
import numpy as np
import pytest
# dig some constants out
FDMDV_OS_48 = codec2.api.FDMDV_OS_48
FDMDV_OS_TAPS_48K = codec2.api.FDMDV_OS_TAPS_48K
FDMDV_OS_TAPS_48_8K = codec2.api.FDMDV_OS_TAPS_48_8K
N8 = 180 # processing buffer size at 8 kHz
N48 = N8 * FDMDV_OS_48 # processing buffer size at 48 kHz
MEM8 = FDMDV_OS_TAPS_48_8K # 8kHz signal filter memory
MEM48 = FDMDV_OS_TAPS_48K # 48kHz signal filter memory
FRAMES = 50 # number of frames to test
FS8 = 8000
FS48 = 48000
AMP = 16000 # sine wave amplitude
FTEST8 = 800 # input test frequency at FS=8kHz
FINTER48 = 10000 # interferer frequency at FS=48kHz
# Due to the design of these resamplers, the processing buffer (at 8kHz)
# must be an integer multiple of oversampling ratio
assert N8 % FDMDV_OS_48 == 0
def test_resampler():
"""
Test for the codec2 audio resampling routine
"""
# time indexes, we advance every frame
t = 0
t1 = 0
# output files to listen to/evaluate result
with open("in8.raw", mode="wb") as fin8:
with open("out48.raw", mode="wb") as f48:
with open("out8.raw", mode="wb") as fout8:
resampler = codec2.resampler()
# Generate FRAMES of a sine wave
for _ in range(FRAMES):
# Primary sine wave, which the down-sampling filter should retain.
sine_in8k = (
AMP * np.cos(2 * np.pi * np.arange(t, t + N8) * FTEST8 / FS8)
).astype(np.int16)
t += N8
sine_in8k.tofile(fin8)
sine_out48k = resampler.resample8_to_48(sine_in8k)
sine_out48k.tofile(f48)
# Add an interfering sine wave, which the down-sampling filter should (mostly) remove
sine_in48k = (
sine_out48k
+ (AMP / 2)
* np.cos(2 * np.pi * np.arange(t1, t1 + N48) * FINTER48 / FS48)
).astype(np.int16)
t1 += N48
sine_out8k = resampler.resample48_to_8(sine_in48k)
sine_out8k.tofile(fout8)
# os.unlink("out48.raw")
# Automated test evaluation --------------------------------------------
# The input and output signals will not be time aligned due to the filter
# delays, so compare the magnitude spectrum
# Read the raw audio files
in8k = np.fromfile("in8.raw", dtype=np.int16)
out8k = np.fromfile("out8.raw", dtype=np.int16)
assert len(in8k) == len(out8k)
# os.unlink("in8.raw")
# os.unlink("out8.raw")
# Apply hanning filter to raw input data samples
h = np.hanning(len(in8k))
S1 = np.abs(np.fft.fft(in8k * h))
S2 = np.abs(np.fft.fft(out8k * h))
# Calculate the ratio between signal and noise (error energy).
error = S1 - S2
error_energy = np.dot(error, error)
ratio = error_energy / np.dot(S1, S1)
ratio_dB = 10 * np.log10(ratio)
# Establish -40.0 as the noise ratio ceiling
threshdB = -40.0
print(f"ratio_dB: {ratio_dB:4.2}" % (ratio_dB))
assert ratio_dB < threshdB
if __name__ == "__main__":
# Run pytest with the current script as the filename.
ecode = pytest.main(["-v", sys.argv[0]])
if ecode == 0:
print("PASS")
else:
print("FAIL")