ca9580cd50
First working version of calibrate.py Replaced analysis parameter dbRefWndMs with dbLinRef rms_analysis.py : Added samples_to_linear_residual() plot_seq_1.py : initial commit.
274 lines
7.8 KiB
Python
274 lines
7.8 KiB
Python
import pprint,queue
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import sounddevice as sd
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import soundfile as sf
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import numpy as np
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from result import Result
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class AudioDevice(object):
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def __init__(self):
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self.inputPortIdx = None
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self.outputPortIdx = None
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self.recordFl = False
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self.sineFl = False
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self.inStream = None
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self.queue = queue.Queue()
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self.bufL = []
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self.bufIdx = -1
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self.srate = 0
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self.ch_cnt = 1
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def setup( self, **kwargs ):
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res = Result()
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if kwargs is None:
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return res
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if 'inPortLabel' in kwargs:
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res += self.select_port( True, kwargs['inPortLabel'] )
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if 'outPortLabel' in kwargs:
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res += self.select_port( False, kwargs['outPortLabel'] )
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return res
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def get_port_label( self, inFl ):
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portL = self.get_port_list(inFl)
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portIdx = self.inputPortIdx if inFl else self.outputPortIdx
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portLabel = None
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if portL and (portIdx is not None):
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for port in portL:
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if portIdx == port['index']:
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portLabel = port['label']
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break
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return portLabel
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def get_state( self ):
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d= {
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'recordFl': self.inStream is not None and self.inStream.active,
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'sineFl': self.sineFl,
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'inPortL': [ d['label'] for d in self.get_port_list(True) ],
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'outPortL': [ d['label'] for d in self.get_port_list(False) ],
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'inPortLabel': self.get_port_label(True),
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'outPortLabel': self.get_port_label(False)
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}
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return d
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def get_port_list( self, inFl ):
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devLabelL = str(sd.query_devices()).split("\n")
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portL = []
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for i,dev in enumerate(sd.query_devices()):
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isInputFl = dev['max_input_channels'] > 0
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isOutputFl = dev['max_output_channels'] > 0
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if (inFl and isInputFl) or (not inFl and isOutputFl):
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portL.append({ 'chN':dev['max_input_channels'], 'label':devLabelL[i].strip(), 'index':i } )
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return portL
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def get_in_port_list( self ):
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return get_port_list( True )
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def get_out_port_list( self ):
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return get_port_list( False )
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def select_port( self, inFl, portLabel ):
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res = Result()
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if inFl and self.inStream is not None and self.inStream.active:
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return res.set_error("A new port may not be selected while it is active." % (portLabel))
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devLabelL = str(sd.query_devices()).split("\n")
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foundFl = False
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if portLabel is None and len(devLabelL)>0:
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portLabel = devLabelL[0]
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portLabel = portLabel.strip()
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N = len(portLabel)
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# for each device
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for i,label in enumerate(devLabelL):
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label = label.strip()
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# if the first N char's of this label match the requested port label
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if len(label)>=N and label[0:N] == portLabel:
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if inFl:
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self.inputPortIdx = i
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else:
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self.outputPortIdx = i
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foundFl = True
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if inFl:
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dev = sd.query_devices()[i]
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self.srate = dev['default_samplerate']
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if self.inStream:
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self.inStream.close()
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self.inStream = sd.InputStream(samplerate=self.srate, device=self.inputPortIdx, channels=self.ch_cnt, callback=self._record_callback, dtype=np.dtype('float32'))
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break
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if not foundFl:
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res.set_error("Unable to locate the audio port named: '%s'." % (portLabel))
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return res
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def is_recording_enabled( self ):
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if self.inStream is None:
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return False
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return self.inStream.active == True
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def record_enable( self, enableFl ):
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# if the input stream has not already been configured
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if enableFl and self.inStream is None:
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return Result(None,'Recording cannot start because a recording input port has not been selected.')
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if not enableFl and self.inStream is None:
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return Result()
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# if the input stream already matches the 'enableFl'.
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if enableFl == self.inStream.active:
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return Result()
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# if we are starting recording
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if enableFl:
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try:
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self.bufL = []
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self.inStream.start()
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except Exception as ex:
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return Result(None,'The recording input stream could not be started. Reason: %s' % str(ex))
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else:
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self.inStream.stop()
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return Result()
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def play_buffer( self ):
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if not self.playFl:
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self.playFl = True
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def play_end( self ):
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if self.playFl:
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self.playFl = False
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def write_buffer( self, fn=None ):
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if fn is None:
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fn = "temp.wav"
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with sf.SoundFile(fn,'w',samplerate=int(self.srate), channels=int(self.ch_cnt)) as f:
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for i,buf in enumerate(self.bufL):
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N = buf.shape[0] if i<len(self.bufL)-1 else self.bufIdx
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f.write(buf[0:N,])
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return Result()
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def buffer_sample_count( self ):
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smpN = 0
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for i in range(len(self.bufL)):
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smpN += self.bufIdx if i == len(self.bufL)-1 else self.bufL[i].shape[0]
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return Result(smpN)
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def buffer_sample_ms( self ):
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r = self.buffer_sample_count()
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if r:
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r.value = int(r.value * 1000.0 / self.srate)
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return r
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def linear_buffer( self ):
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r = self.buffer_sample_count()
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if r:
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smpN = r.value
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bV = np.zeros( (smpN,self.ch_cnt) )
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bi = 0
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for i in range(len(self.bufL)):
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bufSmpN = self.bufIdx if i == len(self.bufL)-1 else self.bufL[i].shape[0]
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bV[ bi:bi+bufSmpN, ] = self.bufL[i][0:bufSmpN]
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bi += bufSmpN
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return Result(bV)
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def _record_callback(self, v, frames, time, status):
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"""This is called (from a separate thread) for each audio block."""
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# send audio to the app
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self.queue.put(v.copy())
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def tick( self, ms ):
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try:
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while True:
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# get audio from the incoming audio thread
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v = self.queue.get_nowait()
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#print(type(v),len(v),np.result_type(v))
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self._update_buffer(v)
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except queue.Empty: # queue was empty
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pass
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return Result()
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def _update_buffer(self,v ):
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# get the length of the last buffer in bufL
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bufN = 0 if len(self.bufL)==0 else self.bufL[-1].shape[0] - self.bufIdx
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# get the length of the incoming sample vector
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vN = v.shape[0]
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# if there are more incoming samples than space in the buffer
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if vN > bufN:
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# store as much of the incoming vector as possible
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if len(self.bufL) > 0:
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self.bufL[-1][self.bufIdx:,:] = v[0:bufN,:]
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vi = bufN # increment the starting position of the src vector
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vN -= bufN # decrement the count of samples that needs to be stored
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# add an empty buffer to bufL[]
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N = int(self.srate * 60) # add a one minute buffer to bufL
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self.bufL.append( np.zeros( (N, self.ch_cnt), dtype=np.dtype('float32') ) )
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self.bufIdx = 0
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self.bufL[-1][self.bufIdx:vN,:] = v[vi:,]
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self.bufIdx += vN
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else:
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self.bufL[-1][self.bufIdx:self.bufIdx+vN,:] = v
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self.bufIdx += vN
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if __name__ == "__main__":
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ad = AudioDevice()
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ad.get_port_list(True)
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