def _cochlear_trim_sai_marginals(filename_and_indexes):
try:
filename, norm_segstart, norm_segend, audio_id, NAP_detail = filename_and_indexes
sai_video_filename = '{}_sai_video_{}'.format(filename, NAP_detail)
if os.path.isfile('{}.npy'.format(sai_video_filename)):
return sai_video_filename
if NAP_detail == 'high':
try:
NAP = utils.csv_to_array(filename + 'cochlear' + NAP_detail)
except:
NAP = brain.cochlear(filename,
stride=1,
rate=44100,
apply_filter=0,
suffix='cochlear' + NAP_detail)
if NAP_detail == 'low':
try:
NAP = utils.csv_to_array(filename + 'cochlear' + NAP_detail)
except:
NAP = brain.cochlear(
filename,
stride=IO.NAP_STRIDE,
rate=IO.NAP_RATE,
apply_filter=0,
suffix='cochlear' + NAP_detail
) # Seems to work best, in particular when they are all the same.
num_channels = NAP.shape[1]
input_segment_width = 2048
sai_params = CreateSAIParams(num_channels=num_channels,
input_segment_width=input_segment_width,
trigger_window_width=input_segment_width,
sai_width=1024)
sai = pysai.SAI(sai_params)
NAP = utils.trim_right(NAP[np.int(np.rint(NAP.shape[0] *
norm_segstart)):np.
int(np.rint(NAP.shape[0] * norm_segend))],
threshold=.05)
sai_video = [
np.copy(sai.RunSegment(input_segment.T))
for input_segment in utils.chunks(NAP, input_segment_width)
]
del NAP
np.save(sai_video_filename,
np.array([sai_rectangles(frame) for frame in sai_video]))
return sai_video_filename
except:
print utils.print_exception(
'Calculation SAI video failed for file {}, NAP detail {}'.format(
filename, NAP_detail))
return False
def main():
x1600 = utils.csv_to_array('data/X1600.csv')
paxes = [] # Array of principal axes for each label
avgs = [] # Average of all the data points for each label
print('Training the classifier using ' + str(np.size(x1600, 1)) +
' points...')
for label in range(0, 10):
print(' Training the digit ' + str(label))
start_col = 1600 * label
end_col = start_col + 1600
data_points = x1600[:, start_col:end_col]
pa, avg = train(data_points, 29)
paxes.append(pa)
avgs.append(avg)
test_points = utils.csv_to_array('data/Te28.csv')
test_labels = utils.csv_to_array('data/Lte28.csv').T
num_points = np.size(test_points, 1)
incorrect = 0 # Number of incorrectly classified points
# Classify each test point and compare it to the real labels
print('Classifying ' + str(num_points) + ' test points...')
for col in range(num_points):
x = test_points[:, col]
errors = [] # The lowest error will correspond to the correct label
# Evaluate how well this data point fits each label
for label in range(10):
# Tensor where columns are principal axis vectors for this label
U = paxes[label]
# Average of all training points that belong to this label
mu = avgs[label]
f = U.T @ (x - mu)
# x_classified represents the data point x, conformed to a label
# The closer x_classified resembles the original data point x,
# the more likely that x belongs to this label
x_classified = (U @ f) + mu
# The smaller this error, the more likely this is our label
e = np.linalg.norm(x - x_classified)
errors.append(e)
best_label = errors.index(min(errors))
if best_label != test_labels[0, col]:
incorrect += 1
print('Misclassified: ' + str(incorrect) + ' out of ' + str(num_points))
def main():
w = utils.csv_to_array(sys.argv[1])
samples = RandomMessage(w)
args = sys.argv[2:]
randomized_msgs = samples.shuffle
# Handle Command-Line Flags
# -export random messages to csv file, each message is a row
if '-export' in args:
utils.array_to_csv(randomized_msgs, "RandomMessage.csv")
# Pass object to any program to use it
return samples
def _cochlear_trim_sai_marginals(filename_and_indexes):
try:
filename, norm_segstart, norm_segend, audio_id, NAP_detail = filename_and_indexes
sai_video_filename = '{}_sai_video_{}'.format(filename, NAP_detail)
if os.path.isfile('{}.npy'.format(sai_video_filename)):
return sai_video_filename
if NAP_detail == 'high':
try:
NAP = utils.csv_to_array(filename+'cochlear'+NAP_detail)
except:
NAP = brain.cochlear(filename, stride=1, rate=44100, apply_filter=0, suffix='cochlear'+NAP_detail)
if NAP_detail == 'low':
try:
NAP = utils.csv_to_array(filename+'cochlear'+NAP_detail)
except:
NAP = brain.cochlear(filename, stride=IO.NAP_STRIDE, rate=IO.NAP_RATE, apply_filter=0, suffix='cochlear'+NAP_detail) # Seems to work best, in particular when they are all the same.
num_channels = NAP.shape[1]
input_segment_width = 2048
sai_params = CreateSAIParams(num_channels=num_channels,
input_segment_width=input_segment_width,
trigger_window_width=input_segment_width,
sai_width=1024)
sai = pysai.SAI(sai_params)
NAP = utils.trim_right(NAP[ np.int(np.rint(NAP.shape[0]*norm_segstart)) : np.int(np.rint(NAP.shape[0]*norm_segend)) ], threshold=.05)
sai_video = [ np.copy(sai.RunSegment(input_segment.T)) for input_segment in utils.chunks(NAP, input_segment_width) ]
del NAP
np.save(sai_video_filename, np.array([ sai_rectangles(frame) for frame in sai_video ]))
return sai_video_filename
except:
print utils.print_exception('Calculation SAI video failed for file {}, NAP detail {}'.format(filename, NAP_detail))
return False
def new_learn_audio(host, debug=False):
context = zmq.Context()
mic = context.socket(zmq.SUB)
mic.connect('tcp://{}:{}'.format(host, IO.MIC))
mic.setsockopt(zmq.SUBSCRIBE, b'')
dreamQ = context.socket(zmq.PUSH)
dreamQ.connect('tcp://{}:{}'.format(host, IO.DREAM))
stateQ, eventQ, brainQ = _three_amigos(context, host)
sender = context.socket(zmq.PUSH)
sender.connect('tcp://{}:{}'.format(host, IO.EXTERNAL))
counterQ = context.socket(zmq.REQ)
counterQ.connect('tcp://{}:{}'.format(host, IO.COUNTER))
poller = zmq.Poller()
poller.register(mic, zmq.POLLIN)
poller.register(stateQ, zmq.POLLIN)
poller.register(eventQ, zmq.POLLIN)
audio = deque()
NAPs = []
wavs = []
wav_audio_ids = {}
NAP_hashes = {}
audio_classifier = []
audio_recognizer = []
global_audio_recognizer = []
mixture_audio_recognizer = []
maxlen = []
deleted_ids = []
state = stateQ.recv_json()
black_list = open('black_list.txt', 'a')
audio_memory = AudioMemory()
if debug:
import matplotlib.pyplot as plt
plt.ion()
while True:
events = dict(poller.poll())
if stateQ in events:
state = stateQ.recv_json()
if mic in events:
new_audio = utils.recv_array(mic)
if state['record']:
audio.append(new_audio)
if eventQ in events:
pushbutton = eventQ.recv_json()
if 'learn' in pushbutton:
try:
t0 = time.time()
filename = pushbutton['filename']
audio_segments = utils.get_segments(filename)
print 'Learning {} duration {} seconds with {} segments'.format(filename, audio_segments[-1], len(audio_segments)-1)
new_sentence = utils.csv_to_array(filename + 'cochlear')
norm_segments = np.rint(new_sentence.shape[0]*audio_segments/audio_segments[-1]).astype('int')
audio_ids = []
new_audio_hash = []
amps = utils.get_amps(filename)
most_significant_value = -np.inf
most_significant_audio_id = []
original_NAP_length = len(NAPs)
for segment, new_sound in enumerate([ utils.trim_right(new_sentence[norm_segments[i]:norm_segments[i+1]]) for i in range(len(norm_segments)-1) ]):
# We filter out short, abrupt sounds with lots of noise.
if np.mean(new_sound) < 2 or new_sound.shape[0] == 0:
black_list.write('{} {}\n'.format(filename, segment))
print 'BLACKLISTED segment {} in file {}'.format(segment, filename)
continue
if debug:
utils.plot_NAP_and_energy(new_sound, plt)
audio_id = audio_memory.learn(new_sound, filename, [ audio_segments[segment], audio_segments[segment+1] ])
# START LEGACY
try:
wavs[audio_id].append(filename)
except:
wavs.append([filename])
wav_audio_ids[(filename, audio_id)] = [ audio_segments[segment], audio_segments[segment+1] ]
# END LEGACY
audio_ids.append(audio_id)
if amps[segment] > most_significant_value:
most_significant_audio_id = audio_id
most_significant_value = amps[segment]
black_list.flush()
print 'AUDIO IDs after blacklisting {}'. format(audio_ids)
if len(audio_ids):
# while len(NAPs) - len(deleted_ids) > AUDIO_MEMORY_SIZE:
# utils.delete_loner(counterQ, NAPs, 'audio_ids_counter', int(AUDIO_MEMORY_SIZE*PROTECT_PERCENTAGE), deleted_ids)
# maxlen = max([ m.shape[0] for memory in NAPs for m in memory if len(m) ])
# memories = [ np.ndarray.flatten(utils.zero_pad(m, maxlen)) for memory in NAPs for m in memory if len(m) ]
# targets = [ i for i,f in enumerate(NAPs) for k in f if len(k) ]
# audio_classifier = train_rPCA_SVM(memories, targets)
# all_hammings = [ utils.hamming_distance(new_audio_hash[i], new_audio_hash[j])
# for i in range(len(new_audio_hash)) for j in range(len(new_audio_hash)) if i > j ]
# print 'RHYME VALUE', np.mean(sorted(all_hammings)[int(len(all_hammings)/2):])
# rhyme = np.mean(sorted(all_hammings)[int(len(all_hammings)/2):]) < RHYME_HAMMERTIME
# sender.send_json('rhyme {}'.format(rhyme))
brainQ.send_pyobj(['audio_learn', filename, audio_ids, audio_memory, most_significant_audio_id, wavs, wav_audio_ids])
print 'Audio learned from {} in {} seconds'.format(filename, time.time() - t0)
else:
print 'SKIPPING fully blacklisted file {}'.format(filename)
except:
utils.print_exception('Audio learning aborted.')
audio.clear()
if 'dream' in pushbutton:
new_dream(audio_memory)
if 'save' in pushbutton:
utils.save('{}.{}'.format(pushbutton['save'], mp.current_process().name), [ deleted_ids, NAPs, wavs, wav_audio_ids, NAP_hashes, audio_classifier, maxlen, audio_memory ])
if 'load' in pushbutton:
deleted_ids, NAPs, wavs, wav_audio_ids, NAP_hashes, audio_classifier, maxlen, audio_memory = utils.load('{}.{}'.format(pushbutton['load'], mp.current_process().name))
def new_respond(control_host, learn_host, debug=False):
context = zmq.Context()
eventQ = context.socket(zmq.SUB)
eventQ.connect('tcp://{}:{}'.format(control_host, IO.EVENT))
eventQ.setsockopt(zmq.SUBSCRIBE, b'')
projector = context.socket(zmq.PUSH)
projector.connect('tcp://{}:{}'.format(control_host, IO.PROJECTOR))
sender = context.socket(zmq.PUSH)
sender.connect('tcp://{}:{}'.format(control_host, IO.EXTERNAL))
brainQ = context.socket(zmq.PULL)
brainQ.bind('tcp://*:{}'.format(IO.BRAIN))
counterQ = context.socket(zmq.REQ)
counterQ.connect('tcp://{}:{}'.format(control_host, IO.COUNTER))
cognitionQ = context.socket(zmq.PUSH)
cognitionQ.connect('tcp://{}:{}'.format(control_host, IO.COGNITION))
association = context.socket(zmq.REQ)
association.connect('tcp://{}:{}'.format(learn_host, IO.ASSOCIATION))
snapshot = context.socket(zmq.REQ)
snapshot.connect('tcp://{}:{}'.format(control_host, IO.SNAPSHOT))
scheduler = context.socket(zmq.PUSH)
scheduler.connect('tcp://{}:{}'.format(control_host, IO.SCHEDULER))
dreamQ = context.socket(zmq.PULL)
dreamQ.bind('tcp://*:{}'.format(IO.DREAM))
snapshot.send_json('Give me state!')
state = snapshot.recv_json()
poller = zmq.Poller()
poller.register(eventQ, zmq.POLLIN)
poller.register(brainQ, zmq.POLLIN)
poller.register(dreamQ, zmq.POLLIN)
sound_to_face = []
wordFace = {}
face_to_sound = []
faceWord = {}
register = {}
video_producer = {}
voiceType1 = 1
voiceType2 = 6
wordSpace1 = 0.3
wordSpaceDev1 = 0.3
wordSpace2 = 0.1
wordSpaceDev2 = 0.3
audio_ids = []
wavs = []
wav_audio_ids = []
NAP_hashes = {}
most_significant_audio_id = []
if debug:
import matplotlib.pyplot as plt
plt.ion()
while True:
events = dict(poller.poll())
if brainQ in events:
cells = brainQ.recv_pyobj()
mode = cells[0]
wav_file = cells[1]
if wav_file not in register:
register[wav_file] = [False, False, False]
if mode == 'audio_learn':
register[wav_file][0] = cells
if mode == 'video_learn':
register[wav_file][1] = cells
if mode == 'face_learn':
register[wav_file][2] = cells
if all(register[wav_file]):
_, _, audio_ids, audio_memory, most_significant_audio_id, wavs, wav_audio_ids = register[wav_file][0]
_, _, tarantino = register[wav_file][1]
_, _, face_id, face_recognizer = register[wav_file][2]
print 'Audio - video - face recognizers related to {} arrived at responder, total processing time {} seconds'.format(wav_file, time.time() - utils.filetime(wav_file))
for audio_id in audio_ids: # If audio_ids is empty, none of this will happen
video_producer[(audio_id, face_id)] = tarantino
if audio_id < len(sound_to_face) and not face_id in sound_to_face[audio_id]: # sound heard before, but not said by this face
sound_to_face[audio_id].append(face_id)
if audio_id == len(sound_to_face):
sound_to_face.append([face_id])
wordFace.setdefault(audio_id, [[face_id,0]])
found = 0
for item in wordFace[audio_id]:
if item[0] == face_id:
item[1] += 1
found = 1
if found == 0:
wordFace[audio_id].append([face_id,1])
# We can't go from a not known face to any of the sounds, that's just the way it is.
print 'face_id for audio segment learned', face_id
if face_id is not -1:
if face_id < len(face_to_sound) and not audio_id in face_to_sound[face_id]: #face seen before, but the sound is new
face_to_sound[face_id].append(audio_id)
if face_id == len(face_to_sound):
face_to_sound.append([audio_id])
faceWord.setdefault(face_id, [[audio_id,0]])
found = 0
for item in faceWord[face_id]:
if item[0] == audio_id:
item[1] += 1
found = 1
if found == 0:
faceWord[face_id].append([audio_id,1])
del register[wav_file]
similar_ids = []
for audio_id in audio_ids:
# I SUSPECT THIS IS WRONG, SINCE THERE IS NO SORTING OF THESE HAMMING DISTANCES IN ASSOCIATION.PY
new_audio_hash = audio_memory.audio_ids[audio_id][-1].crude_hash
similar_ids_for_this_audio_id = [ utils.hamming_distance(new_audio_hash, random.choice(h).crude_hash) for h in audio_memory.audio_ids.itervalues() ]
similar_ids.append(similar_ids_for_this_audio_id)
if len(audio_ids):
association.send_pyobj(['analyze',wav_file,wav_audio_ids,audio_ids,wavs,similar_ids,wordFace,faceWord])
association.recv_pyobj()
sender.send_json('last_most_significant_audio_id {}'.format(most_significant_audio_id))
cognitionQ.send_pyobj(face_recognizer) # A possiblity of recognizing a face that is not connecting to any soundfiles
if eventQ in events:
pushbutton = eventQ.recv_json()
if 'respond_single' in pushbutton:
try:
filename = pushbutton['filename']
audio_segments = utils.get_segments(filename)
print 'Single response to {} duration {} seconds with {} segments'.format(filename, audio_segments[-1], len(audio_segments)-1)
new_sentence = utils.csv_to_array(filename + 'cochlear')
norm_segments = np.rint(new_sentence.shape[0]*audio_segments/audio_segments[-1]).astype('int')
segment_id = utils.get_most_significant_word(filename)
NAP = utils.trim_right(new_sentence[norm_segments[segment_id]:norm_segments[segment_id+1]])
if debug:
plt.imshow(NAP.T, aspect='auto')
plt.draw()
best_match,_,_,_,_ = audio_memory.find(NAP)
soundfile = best_match.wav_file
segstart, segend = best_match.segment_idxs
voiceChannel = 1
speed = 1
amp = -3 # voice amplitude in dB
_,dur,maxamp,_ = utils.getSoundInfo(soundfile)
start = 0
voice1 = 'playfile {} {} {} {} {} {} {} {} {}'.format(1, voiceType1, start, soundfile, speed, segstart, segend, amp, maxamp)
voice2 = ''
print 'Recognized as sound {}'.format(best_match.audio_id)
# sound_to_face, video_producer
projection = _project(best_match.audio_id, sound_to_face, NAP, video_producer)
scheduler.send_pyobj([[ dur, voice1, voice2, projection, FRAME_SIZE ]])
print 'Respond time from creation of wav file was {} seconds'.format(time.time() - utils.filetime(filename))
except:
utils.print_exception('Single response aborted.')
if 'play_sentence' in pushbutton:
try:
sentence = pushbutton['sentence']
sentence = eval(sentence)
print '*** (play) Play sentence', sentence
start = 0
nextTime1 = 0
play_events = []
for i in range(len(sentence)):
word_id = sentence[i]
soundfile = np.random.choice(wavs[word_id])
speed = 1
segstart, segend = wav_audio_ids[(soundfile, word_id)]
NAP = _extract_NAP(segstart, segend, soundfile)
amp = -3 # voice amplitude in dB
_,totaldur,maxamp,_ = utils.getSoundInfo(soundfile)
dur = segend-segstart
if dur <= 0: dur = totaldur
# play in both voices
voice1 = 'playfile {} {} {} {} {} {} {} {} {}'.format(1, voiceType1, start, soundfile, speed, segstart, segend, amp, maxamp)
voice2 = 'playfile {} {} {} {} {} {} {} {} {}'.format(2, voiceType1, start, soundfile, speed, segstart, segend, amp, maxamp)
wordSpacing1 = wordSpace1 + np.random.random()*wordSpaceDev1
print 'PLAY RESPOND SPACING', wordSpacing1
nextTime1 += (dur/speed)+wordSpacing1
projection = _project(audio_id, sound_to_face, NAP, video_producer)
play_events.append([ dur+wordSpacing1, voice1, voice2, projection, FRAME_SIZE ])
scheduler.send_pyobj(play_events)
except:
utils.print_exception('Sentence play aborted.')
if 'respond_sentence' in pushbutton:
print 'SENTENCE Respond to', pushbutton['filename'][-12:]
try:
filename = pushbutton['filename']
audio_segments = utils.get_segments(filename)
print 'Sentence response to {} duration {} seconds with {} segments'.format(filename, audio_segments[-1], len(audio_segments)-1)
new_sentence = utils.csv_to_array(filename + 'cochlear')
norm_segments = np.rint(new_sentence.shape[0]*audio_segments/audio_segments[-1]).astype('int')
segment_id = utils.get_most_significant_word(filename)
print '**Sentence selected to respond to segment {}'.format(segment_id)
NAP = utils.trim_right(new_sentence[norm_segments[segment_id]:norm_segments[segment_id+1]])
best_match,_,_,_,_ = audio_memory.find(NAP)
audio_id = best_match.audio_id
soundfile = best_match.wav_file
numWords = len(audio_segments)-1
print numWords
association.send_pyobj(['setParam', 'numWords', numWords ])
association.recv_pyobj()
association.send_pyobj(['makeSentence', audio_id])
print 'respond_sentence waiting for association output...',
sentence, secondaryStream = association.recv_pyobj()
print '*** (respond) Play sentence', sentence, secondaryStream
start = 0
nextTime1 = 0
nextTime2 = 0
enableVoice2 = 1
play_events = []
for i in range(len(sentence)):
word_id = sentence[i]
soundfile = np.random.choice(wavs[word_id])
voiceChannel = 1
speed = 1
# segment start and end within sound file, if zero, play whole file
segstart, segend = wav_audio_ids[(soundfile, word_id)]
NAP = _extract_NAP(segstart, segend, soundfile)
amp = -3 # voice amplitude in dB
#totaldur, maxamp = utils.getSoundParmFromFile(soundfile)
_,totaldur,maxamp,_ = utils.getSoundInfo(soundfile)
dur = segend-segstart
if dur <= 0: dur = totaldur
voice1 = 'playfile {} {} {} {} {} {} {} {} {}'.format(voiceChannel, voiceType1, start, soundfile, speed, segstart, segend, amp, maxamp)
#start += dur # if we want to create a 'score section' for Csound, update start time to make segments into a contiguous sentence
wordSpacing1 = wordSpace1 + np.random.random()*wordSpaceDev1
nextTime1 += (dur/speed)+wordSpacing1
#print 'voice 2 ready to play', secondaryStream[i], i
if enableVoice2:
word_id2 = secondaryStream[i]
#print 'voice 2 playing', secondaryStream[i]
soundfile2 = np.random.choice(wavs[word_id2])
voiceChannel2 = 2
start2 = 0.7 # set delay between voice 1 and 2
speed2 = 0.7
amp2 = -10 # voice amplitude in dB
try:
segstart2, segend2 = wav_audio_ids[(soundfile2, word_id2)]
dur2 = segend2-segstart2
#totalDur2, maxamp2 = utils.getSoundParmFromFile(soundfile2)
_,totalDur2,maxamp2,_ = utils.getSoundInfo(soundfile)
if dur2 <= 0: dur2 = totalDur2
voice2 = 'playfile {} {} {} {} {} {} {} {} {}'.format(voiceChannel2, voiceType2, start2, soundfile2, speed2, segstart2, segend2, amp2, maxamp2)
wordSpacing2 = wordSpace2 + np.random.random()*wordSpaceDev2
nextTime2 += (dur2/speed2)+wordSpacing2
except:
voice2 = ''
utils.print_exception('VOICE 2 tried to access an illegal soundfile/audio_id combination.')
#enableVoice2 = 0
# trig another word in voice 2 only if word 2 has finished playing (and sync to start of voice 1)
if nextTime1 > nextTime2: enableVoice2 = 1
projection = _project(audio_id, sound_to_face, NAP, video_producer)
print 'SENTENCE RESPOND SPACING', wordSpacing1
play_events.append([ dur+wordSpacing1, voice1, voice2, projection, FRAME_SIZE ])
scheduler.send_pyobj(play_events)
print 'Sentence respond time from creation of wav file was {} seconds'.format(time.time() - utils.filetime(filename))
except:
utils.print_exception('Sentence response aborted.')
if 'testSentence' in pushbutton:
print 'testSentence', pushbutton
association.send_pyobj(['makeSentence',int(pushbutton['testSentence'])])
print 'testSentence waiting for association output...'
sentence, secondaryStream = association.recv_pyobj()
print '*** Test sentence', sentence, secondaryStream
if 'assoc_setParam' in pushbutton:
try:
parm, value = pushbutton['assoc_setParam'].split()
association.send_pyobj(['setParam', parm, value ])
association.recv_pyobj()
except:
utils.print_exception('Assoc set param aborted.')
if 'respond_setParam' in pushbutton:
items = pushbutton['respond_setParam'].split()
if items[0] == 'voiceType':
chan = items[1]
if chan == '1': voiceType1 = int(items[2])
if chan == '2': voiceType2 = int(items[2])
if items[0] == 'wordSpace':
chan = items[1]
print 'wordSpace chan', chan, items
if chan == '1': wordSpace1 = float(items[2])
if chan == '2': wordSpace2 = float(items[2])
if items[0] == 'wordSpaceDev':
chan = items[1]
print 'wordSpaceDev1 chan', chan, items
if chan == '1': wordSpaceDev1 = float(items[2])
if chan == '2': wordSpaceDev2 = float(items[2])
if 'play_id' in pushbutton:
try:
items = pushbutton['play_id'].split(' ')
if len(items) < 3: print 'PARAMETER ERROR: play_id audio_id voiceChannel voiceType'
play_audio_id = int(items[0])
voiceChannel = int(items[1])
voiceType = int(items[2])
print 'play_audio_id', play_audio_id, 'voice', voiceChannel
print 'wavs[play_audio_id]', wavs[play_audio_id]
#print wavs
soundfile = np.random.choice(wavs[play_audio_id])
speed = 1
#print 'wav_audio_ids', wav_audio_ids
segstart, segend = wav_audio_ids[(soundfile, play_audio_id)]
#segstart = 0 # segment start and end within sound file
#segend = 0 # if zero, play whole file
amp = -3 # voice amplitude in dB
#dur, maxamp = utils.getSoundParmFromFile(soundfile)
_,dur,maxamp,_ = utils.getSoundInfo(soundfile)
start = 0
sender.send_json('playfile {} {} {} {} {} {} {} {} {}'.format(voiceChannel, voiceType, start, soundfile, speed, segstart, segend, amp, maxamp))
except:
utils.print_exception('play_id aborted.')
if 'print_me' in pushbutton:
# just for inspecting the contents of objects while running
print 'printing '+pushbutton['print_me']
if 'brain ' in pushbutton['print_me']:
print_variable = pushbutton['print_me'].split('brain ')[-1]
try:
print eval(print_variable)
except Exception, e:
print e, 'print_me in brain failed.'
elif 'association ' in pushbutton['print_me']:
print_variable = pushbutton['print_me'].split('association ')[-1]
association.send_pyobj(['print_me',print_variable])
if 'dream' in pushbutton:
play_events = []
for audio_segment in audio_memory.all_segments():
segstart, segend = audio_segment.segment_idxs
dur = segend - segstart
NAP = _extract_NAP(segstart, segend, audio_segment.wav_file)
speed = 1
amp = -3
maxamp = 1
start = 0
voice1 = 'playfile {} {} {} {} {} {} {} {} {}'.format(1, 6, np.random.rand()/3, audio_segment.wav_file, speed, segstart, segend, amp, maxamp)
projection = _project(audio_segment.audio_id, sound_to_face, NAP, video_producer)
voice2 = 'playfile {} {} {} {} {} {} {} {} {}'.format(2, 6, np.random.randint(3,6), audio_segment.wav_file, speed, segstart, segend, amp, maxamp)
play_events.append([ dur, voice1, voice2, projection, FRAME_SIZE ])
print 'Dream mode playing back {} memories'.format(len(play_events))
scheduler.send_pyobj(play_events)
if 'save' in pushbutton:
utils.save('{}.{}'.format(pushbutton['save'], mp.current_process().name), [ sound_to_face, wordFace, face_to_sound, faceWord, video_producer, wavs, wav_audio_ids, audio_classifier, maxlen, NAP_hashes, face_id, face_recognizer, audio_memory ])
if 'load' in pushbutton:
sound_to_face, wordFace, face_to_sound, faceWord, video_producer, wavs, wav_audio_ids, audio_classifier, maxlen, NAP_hashes, face_id, face_recognizer, audio_memory = utils.load('{}.{}'.format(pushbutton['load'], mp.current_process().name))
