def make_noise(text_length, filename):
# checking audio file to match with
nb_samples = audiofile.samples(filename)
sampling_rate = audiofile.sampling_rate(filename) # in Hz
# Generate random logarithmic noise
noise = np.random.lognormal(0, 1, nb_samples)
noise /= np.amax(np.abs(noise))
# Filter with bandpass filter
nyq = 0.5 * sampling_rate
low = text_length / nyq
high = (50 + text_length) / nyq
order = 1
b, a = signal.butter(order, [low, high], btype="band")
filtered_noise = signal.lfilter(b, a, noise)
# create Gaussian enveloppe
t = np.linspace(start=-0.5, stop=0.5, num=nb_samples, endpoint=False)
i, e = signal.gausspulse(t, fc=5, retenv=True, bwr=-6.0)
out_signal = np.multiply(filtered_noise, e)
out_signal *= 30
# write audio file
audiofile.write(filename + ".noise.wav", out_signal, sampling_rate)
print("text length was :", text_length)
def test_call(tmpdir, bit_depth_in, channels_in, format_in, sampling_rate_in,
flavor, bit_depth_out, channels_out, format_out,
sampling_rate_out):
file_in = os.path.join(tmpdir, 'in.' + format_in)
file_out = os.path.join(tmpdir, 'out.' + format_out)
signal = np.zeros((channels_in, int(sampling_rate_in)), np.float32)
if format_in == 'mp3':
audiofile.write(
f'{file_in[:-4]}.wav',
signal,
sampling_rate_in,
int(bit_depth_in),
)
os.rename(f'{file_in[:-4]}.wav', file_in)
else:
audiofile.write(
file_in,
signal,
int(sampling_rate_in),
int(bit_depth_in),
)
flavor(file_in, file_out)
assert audiofile.bit_depth(file_out) == bit_depth_out
assert audiofile.channels(file_out) == channels_out
assert audiofile.sampling_rate(file_out) == sampling_rate_out
def deduce_properties(self):
# Audio file duration
self.duration = af.duration(self.file.path)
# Audio file sampling rate
rate = af.sampling_rate(self.file.path)
self._deduce_tempo(rate)
self.save()
def _add_media(
self,
root: str,
file: str,
version: str,
archive: str = None,
checksum: str = None,
):
r"""Add or update media file.
If you want to update only the version
of an unaltered media file,
don't specify ``archive`` and ``checksum``.
Args:
root: root directory
file: relative file path
archive: archive name without extension
checksum: checksum of file
version: version string
"""
format = audeer.file_extension(file).lower()
if archive is None:
archive = self.archive(file)
if checksum is None:
checksum = self.checksum(file)
bit_depth = self.bit_depth(file)
channels = self.channels(file)
duration = self.duration(file)
sampling_rate = self.sampling_rate(file)
else:
bit_depth = channels = sampling_rate = 0
duration = 0.0
if format in define.FORMATS:
path = os.path.join(root, file)
bit_depth = audiofile.bit_depth(path)
channels = audiofile.channels(path)
duration = audiofile.duration(path)
sampling_rate = audiofile.sampling_rate(path)
self._df.loc[file] = [
archive,
bit_depth,
channels,
checksum,
duration,
format,
0, # removed
sampling_rate,
define.DependType.MEDIA,
version,
]
def deduce_properties(self):
'''
Deduces the sample duration and tempo.
'''
# Duration
self.duration = af.duration(self.file.path)
# Sampling rate
rate = af.sampling_rate(self.file.path)
self._deduce_tempo(rate)
self.save()
def test_broken_file(non_audio_file):
# Only match the beginning of error message
# as the default soundfile message differs at the end on macOS
error_msg = 'Error opening'
# Reading file
with pytest.raises(RuntimeError, match=error_msg):
af.read(non_audio_file)
# Metadata
if audeer.file_extension(non_audio_file) == 'wav':
with pytest.raises(RuntimeError, match=error_msg):
af.bit_depth(non_audio_file)
else:
assert af.bit_depth(non_audio_file) is None
with pytest.raises(RuntimeError, match=error_msg):
af.channels(non_audio_file)
with pytest.raises(RuntimeError, match=error_msg):
af.duration(non_audio_file)
with pytest.raises(RuntimeError, match=error_msg):
af.samples(non_audio_file)
with pytest.raises(RuntimeError, match=error_msg):
af.sampling_rate(non_audio_file)
def test_empty_file(empty_file):
# Reading file
signal, sampling_rate = af.read(empty_file)
assert len(signal) == 0
# Metadata
for sloppy in [True, False]:
assert af.duration(empty_file, sloppy=sloppy) == 0.0
assert af.channels(empty_file) == 1
assert af.sampling_rate(empty_file) == sampling_rate
assert af.samples(empty_file) == 0
if audeer.file_extension(empty_file) == 'wav':
assert af.bit_depth(empty_file) == 16
else:
assert af.bit_depth(empty_file) is None
def test_sampling_rate(sampling_rate):
db = audb.load(
DB_NAME,
sampling_rate=sampling_rate,
full_path=False,
num_workers=pytest.NUM_WORKERS,
verbose=False,
)
original_files = db['files']['original'].get()
df = audb.cached()
assert df['sampling_rate'].values[0] == sampling_rate
for converted_file, original_file in zip(db.files, original_files):
converted_file = os.path.join(db.meta['audb']['root'], converted_file)
original_file = os.path.join(DB_ROOT, original_file)
if sampling_rate is None:
assert audiofile.sampling_rate(converted_file) == \
audiofile.sampling_rate(original_file)
else:
assert audiofile.sampling_rate(converted_file) == sampling_rate
def test_mp3(tmpdir, magnitude, sampling_rate, channels):
# Currently we are not able to setup the Windows runner with MP3 support
# https://github.com/audeering/audiofile/issues/51
if sys.platform == 'win32':
return
signal = sine(magnitude=magnitude,
sampling_rate=sampling_rate,
channels=channels)
# Create wav file and use sox to convert to mp3
wav_file = str(tmpdir.join('signal.wav'))
mp3_file = str(tmpdir.join('signal.mp3'))
af.write(wav_file, signal, sampling_rate)
subprocess.call(['sox', wav_file, mp3_file])
assert audeer.file_extension(mp3_file) == 'mp3'
sig, fs = af.read(mp3_file)
assert_allclose(_magnitude(sig), magnitude, rtol=0, atol=tolerance(16))
assert fs == sampling_rate
assert _channels(sig) == channels
if channels == 1:
assert sig.ndim == 1
else:
assert sig.ndim == 2
assert af.channels(mp3_file) == _channels(sig)
assert af.sampling_rate(mp3_file) == sampling_rate
assert af.samples(mp3_file) == _samples(sig)
assert af.duration(mp3_file) == _duration(sig, sampling_rate)
assert af.duration(mp3_file,
sloppy=True) == sox.file_info.duration(mp3_file)
assert af.bit_depth(mp3_file) is None
# Test additional arguments to read with sox
offset = 0.1
duration = 0.5
sig, fs = af.read(mp3_file, offset=offset, duration=duration)
assert _duration(sig, sampling_rate) == duration
sig, fs = af.read(mp3_file, offset=offset)
# Don't test for 48000 Hz and 2 channels
# https://github.com/audeering/audiofile/issues/23
if not (sampling_rate == 48000 and channels == 2):
assert_allclose(
_duration(sig, sampling_rate),
af.duration(mp3_file) - offset,
rtol=0,
atol=tolerance('duration', sampling_rate),
)
def _check_convert(
self,
file: str,
bit_depth: typing.Optional[int],
channels: typing.Optional[int],
sampling_rate: typing.Optional[int],
) -> bool:
r"""Check if file needs to be converted to flavor."""
format = audeer.file_extension(file).lower()
# format change
if self.format is not None:
if self.format != format:
return True
convert = False
# precision change
if not convert and self.bit_depth is not None:
bit_depth = bit_depth or audiofile.bit_depth(file)
if self.bit_depth != bit_depth:
convert = True
# mixdown and channel selection
if not convert and self.mixdown or self.channels is not None:
channels = channels or audiofile.channels(file)
if self.mixdown and channels != 1:
convert = True
elif list(range(channels)) != self.channels:
convert = True
# sampling rate change
if not convert and self.sampling_rate is not None:
sampling_rate = sampling_rate or audiofile.sampling_rate(file)
if self.sampling_rate != sampling_rate:
convert = True
if convert and format not in define.FORMATS:
raise RuntimeError(
f"You have to specify the 'format' argument "
f"to convert '{file}' "
f"to the specified flavor "
f"as we cannot write to {format.upper()} files."
)
return convert
def test_formats():
files = [
'gs-16b-1c-44100hz.opus',
'gs-16b-1c-8000hz.amr',
'gs-16b-1c-44100hz.m4a',
'gs-16b-1c-44100hz.aac',
]
header_durations = [ # as given by mediainfo
15.839,
15.840000,
15.833,
None,
]
files = [os.path.join(ASSETS_DIR, f) for f in files]
for file, header_duration in zip(files, header_durations):
signal, sampling_rate = af.read(file)
assert af.channels(file) == _channels(signal)
assert af.sampling_rate(file) == sampling_rate
assert af.samples(file) == _samples(signal)
duration = _duration(signal, sampling_rate)
assert af.duration(file) == duration
if header_duration is None:
# Here we expect samplewise precision
assert af.duration(file, sloppy=True) == duration
else:
# Here we expect limited precision
# as the results differ between soxi and mediainfo
precision = 1
sloppy_duration = round(af.duration(file, sloppy=True), precision)
header_duration = round(header_duration, precision)
assert sloppy_duration == header_duration
assert af.bit_depth(file) is None
if file.endswith('m4a'):
# Test additional arguments to read with ffmpeg
offset = 0.1
duration = 0.5
sig, fs = af.read(file, offset=offset, duration=duration)
assert _duration(sig, sampling_rate) == duration
sig, fs = af.read(file, offset=offset)
assert _duration(sig, sampling_rate) == af.duration(file) - offset
import numpy as np from scipy import signal import audiofile # coming from generate.py text_length = 5000 filmename = "test_1.wav" # checking files nb_samples = audiofile.samples(filmename) sampling_rate = audiofile.sampling_rate(filmename) # in Hz # Generate random noise noise = np.random.lognormal(0, 1, nb_samples) noise /= np.amax(np.abs(noise)) # Filter with bandpass filter nyq = 0.5 * sampling_rate low = text_length / nyq high = (50 + text_length) / nyq order = 1 b, a = signal.butter(order, [low, high], btype='band') filtered_noise = signal.lfilter(b, a, noise) # create Gaussian enveloppe #t = np.linspace(start=-0.5,stop=0.5, num=nb_samples, endpoint=False) #i, e = signal.gausspulse(t, fc=5, retenv=True, bwr=-6.0) #out_signal = np.multiply(filtered_noise, e) #out_signal *= 0.3 #out_signal = i
def test_publish(version):
db = audformat.Database.load(DB_ROOT_VERSION[version])
print(db.is_portable)
print(db.files)
if not audb.versions(DB_NAME):
with pytest.raises(RuntimeError):
audb.latest_version(DB_NAME)
archives = db['files']['speaker'].get().dropna().to_dict()
deps = audb.publish(
DB_ROOT_VERSION[version],
version,
pytest.PUBLISH_REPOSITORY,
archives=archives,
previous_version=None,
num_workers=pytest.NUM_WORKERS,
verbose=False,
)
backend = audb.core.utils.lookup_backend(DB_NAME, version)
number_of_files = len(set(archives.keys()))
number_of_archives = len(set(archives.values()))
assert len(deps.files) - len(deps.archives) == (number_of_files -
number_of_archives)
for archive in set(archives.values()):
assert archive in deps.archives
db = audb.load(
DB_NAME,
version=version,
full_path=False,
num_workers=pytest.NUM_WORKERS,
)
assert db.name == DB_NAME
versions = audb.versions(DB_NAME)
latest_version = audb.latest_version(DB_NAME)
assert version in versions
assert latest_version == versions[-1]
df = audb.available(only_latest=False)
assert DB_NAME in df.index
assert set(df[df.index == DB_NAME]['version']) == set(versions)
df = audb.available(only_latest=True)
assert DB_NAME in df.index
assert df[df.index == DB_NAME]['version'][0] == latest_version
for file in db.files:
name = archives[file] if file in archives else file
file_path = backend.join(db.name, 'media', name)
backend.exists(file_path, version)
path = os.path.join(DB_ROOT_VERSION[version], file)
assert deps.checksum(file) == audbackend.md5(path)
if deps.format(file) in [
audb.core.define.Format.WAV,
audb.core.define.Format.FLAC,
]:
assert deps.bit_depth(file) == audiofile.bit_depth(path)
assert deps.channels(file) == audiofile.channels(path)
assert deps.duration(file) == audiofile.duration(path)
assert deps.sampling_rate(file) == audiofile.sampling_rate(path)
def test_wav(tmpdir, bit_depth, duration, sampling_rate, channels, always_2d):
file = str(tmpdir.join('signal.wav'))
signal = sine(duration=duration,
sampling_rate=sampling_rate,
channels=channels)
sig, fs = write_and_read(
file,
signal,
sampling_rate,
bit_depth=bit_depth,
always_2d=always_2d,
)
# Expected number of samples
samples = int(np.ceil(duration * sampling_rate))
# Compare with sox implementation to check write()
assert_allclose(sox.file_info.duration(file),
duration,
rtol=0,
atol=tolerance('duration', sampling_rate))
assert sox.file_info.sample_rate(file) == sampling_rate
assert sox.file_info.channels(file) == channels
assert sox.file_info.num_samples(file) == samples
assert sox.file_info.bitdepth(file) == bit_depth
# Compare with signal values to check read()
assert_allclose(_duration(sig, fs),
duration,
rtol=0,
atol=tolerance('duration', sampling_rate))
assert fs == sampling_rate
assert _channels(sig) == channels
assert _samples(sig) == samples
# Test audiofile metadata methods
assert_allclose(af.duration(file),
duration,
rtol=0,
atol=tolerance('duration', sampling_rate))
assert af.sampling_rate(file) == sampling_rate
assert af.channels(file) == channels
assert af.samples(file) == samples
assert af.bit_depth(file) == bit_depth
# Test types of audiofile metadata methods
assert type(af.duration(file)) is float
assert type(af.sampling_rate(file)) is int
assert type(af.channels(file)) is int
assert type(af.samples(file)) is int
# Test dimensions of array
if channels == 1 and not always_2d:
assert sig.ndim == 1
else:
assert sig.ndim == 2
# Test additional arguments to read
if sampling_rate > 100:
offset = 0.001
duration = duration - 2 * offset
sig, fs = af.read(
file,
offset=offset,
duration=duration,
always_2d=always_2d,
)
assert _samples(sig) == int(np.ceil(duration * sampling_rate))