def test_Save(self):
cidx = self._sample_1.extract_channel(0)
channel = self._sample_1.get_channel(cidx)
audio = Audio()
audio.append_channel( channel )
signals.save( TestChannel._sample_path_new, audio )
savedaudio = signals.open(TestChannel._sample_path_new)
self._sample_1.rewind()
frames = self._sample_1.read_frames( self._sample_1.get_nframes() )
savedframes = savedaudio.read_frames( self._sample_1.get_nframes() )
self.assertEqual(len(frames), len(savedframes))
self.assertEqual(frames, savedframes)
def export(self, inputname, outputname):
"""
Create a new wav file with requested parameters.
@param inputname (string) name of the inputfile
@param reqSamplewidth (string) name of the outputfile
"""
toconvert = False
audio = signals.open(inputname)
if (audio.get_sampwidth() < self._reqSamplewidth):
raise NameError("The sample width of ("+str(audio.get_sampwidth())+") of the given file is not appropriate. " + str(self._reqSamplewidth) + " bytes required")
if (audio.get_framerate() < self._reqFramerate):
raise NameError("The framerate of ("+str(audio.get_framerate())+") of the given file is not appropriate. " + str(self._reqFramerate) + " Hz required")
if (self._reqSamplewidth != audio.get_sampwidth()):
toconvert = True
if self._logfile:
self._logfile.print_message("The sample width of ("+str(audio.get_sampwidth())+") of the given file is not appropriate. Sample width is changed to " + str(self._reqSamplewidth) + " bytes", indent=3, status=1)
if (self._reqChannels != audio.get_nchannels()):
toconvert = True
if self._logfile:
self._logfile.print_message("The number of channels of ("+str(audio.get_nchannels())+") of the given file is not appropriate. Number of channels is changed to " + str(self._reqChannels) + " channels", indent=3, status=1)
if (self._reqFramerate != audio.get_framerate()):
toconvert = True
if self._logfile:
self._logfile.print_message("The framerate of ("+str(audio.get_framerate())+") of the given file is not appropriate. Framerate is changed to " + str(self._reqFramerate) + " Hz", indent=3, status=1)
if toconvert is True:
# Get the expected channel
idx = audio.extract_channel(0)
# no more need of input data, can close
audio.close()
# Do the job (do not modify the initial channel).
formatter = ChannelFormatter( audio.get_channel(idx) )
formatter.set_framerate(self._reqFramerate)
formatter.set_sampwidth(self._reqSamplewidth)
formatter.convert()
# Save the converted channel
audio_out = Audio()
audio_out.append_channel( formatter.channel )
signals.save( outputname, audio_out )
return toconvert
def run(self, audiofile, trsfile, output_dir, output_ext="TextGrid"):
"""
Split an audio file into multiple small audio file.
@param audiofile is the audio input file name
@param trsfile is the transcription input file name
@param output_dir is a directory name to save output tracks (one per unit)
@param output_ext (default TextGrid)
"""
if not os.path.exists(output_dir):
os.mkdir(output_dir)
audiospeech = signals.open(audiofile)
transcription = annotationdata.io.read(trsfile)
tracks_tier = None
for tier in transcription:
if "name" in tier.GetName().lower():
tracks_tier = tier
if tracks_tier is None:
raise Exception("Expected tier not found: a tier name must contain 'name'")
list_transcription = TrsUtils.Split(transcription, tracks_tier)
names = [a.GetLabel().GetValue() for a in tracks_tier if not a.GetLabel().IsEmpty()]
trstracks = []
for trs in list_transcription:
begin = int(trs.GetBegin() * audiospeech.get_framerate())
end = int(trs.GetEnd() * audiospeech.get_framerate())
trstracks.append((begin, end))
TrsUtils.Shift(trs, trs.GetBegin())
chunks = []
nframes = audiospeech.get_nframes()
for from_pos, to_pos in trstracks:
if nframes < from_pos:
raise ValueError("Position %d not in range(%d)" % (from_pos, nframes))
audiospeech.set_pos(from_pos)
chunks.append(audiospeech.read_frames(to_pos - from_pos))
for name, chunk, trs in zip(names, chunks, list_transcription):
signals.save(os.path.join(output_dir, name + ".wav"), chunk)
annotationdata.io.write(os.path.join(output_dir, name + "." + output_ext), trs)
def test_WriteFrames(self):
_sample_new = "newFile.wav"
# save first
signals.save( _sample_new, self._sample_1 )
# read the saved file and compare Audio() instances
newFile = signals.open( _sample_new )
self.assertEqual(newFile.get_framerate(), self._sample_1.get_framerate())
self.assertEqual(newFile.get_sampwidth(), self._sample_1.get_sampwidth())
self.assertEqual(newFile.get_nchannels(), self._sample_1.get_nchannels())
self.assertEqual(newFile.get_nframes(), self._sample_1.get_nframes())
newFile.close()
os.remove(_sample_new)
self._sample_1.rewind()
signals.save_fragment( _sample_new, self._sample_1, self._sample_1.read_frames(self._sample_1.get_nframes()))
newFile = signals.open( _sample_new )
self.assertEqual(newFile.get_framerate(), self._sample_1.get_framerate())
self.assertEqual(newFile.get_sampwidth(), self._sample_1.get_sampwidth())
self.assertEqual(newFile.get_nchannels(), self._sample_1.get_nchannels())
self.assertEqual(newFile.get_nframes(), self._sample_1.get_nframes())
newFile.close()
os.remove(_sample_new)
_sample_new = "newFile.aiff"
# save first
signals.save( _sample_new, self._sample_4 )
# read the saved file and compare Audio() instances
newFile = signals.open( _sample_new )
self.assertEqual(newFile.get_framerate(), self._sample_4.get_framerate())
self.assertEqual(newFile.get_sampwidth(), self._sample_4.get_sampwidth())
self.assertEqual(newFile.get_nchannels(), self._sample_4.get_nchannels())
self.assertEqual(newFile.get_nframes(), self._sample_4.get_nframes())
newFile.close()
os.remove(_sample_new)
self._sample_4.rewind()
signals.save_fragment( _sample_new, self._sample_4, self._sample_4.read_frames(self._sample_4.get_nframes()))
newFile = signals.open( _sample_new )
self.assertEqual(newFile.get_framerate(), self._sample_4.get_framerate())
self.assertEqual(newFile.get_sampwidth(), self._sample_4.get_sampwidth())
self.assertEqual(newFile.get_nchannels(), self._sample_4.get_nchannels())
self.assertEqual(newFile.get_nframes(), self._sample_4.get_nframes())
newFile.close()
os.remove(_sample_new)
if args.bf and args.bs:
print "bf option and bs option can't be used at the same time !"
sys.exit(1)
if args.ef and args.es:
print "ef option and es option can't be used at the same time !"
sys.exit(1)
if args.bf:
begin = args.bf
elif args.bs:
begin = args.bs*audio.get_framerate()
else:
begin = 0
if args.ef:
end = args.ef
elif args.es:
end = args.es*audio.get_framerate()
else:
end = 0
for i in xrange(audio.get_nchannels()):
idx = audio.extract_channel(i)
audio.rewind()
channel = audio.get_channel(idx)
extracter = ChannelFragmentExtracter(channel)
audio_out.append_channel(extracter.extract_fragment(begin, end))
signals.save(args.o, audio_out)
# ----------------------------------------------------------------------------
equalizer = ChannelsEqualizer()
file = signals.open(args.w[0])
sampwidth = file.get_sampwidth()
framerate = file.get_framerate()
for inputFile in args.w:
audio = signals.open(inputFile)
if audio.get_sampwidth() != sampwidth:
print "Input files must have the same sample width !"
sys.exit(1)
if audio.get_framerate() != framerate:
print "Input files must have the same framerate !"
sys.exit(1)
idx = audio.extract_channel(1)
equalizer.append_channel(audio.get_channel(idx))
equalizer.equalize()
# Save the converted channel
for i, chan in enumerate(equalizer.channels):
audio_out = Audio()
audio_out.append_channel( chan )
filename, extension = os.path.splitext(args.w[i])
signals.save(filename + "EQUAL" + extension, audio_out)
# ----------------------------------------------------------------------------