本文整理汇总了Python中wave.open_audio函数的典型用法代码示例。如果您正苦于以下问题：Python open_audio函数的具体用法？Python open_audio怎么用？Python open_audio使用的例子？那么恭喜您, 这里精选的函数代码示例或许可以为您提供帮助。
在下文中一共展示了open_audio函数的15个代码示例，这些例子默认根据受欢迎程度排序。您可以为喜欢或者感觉有用的代码点赞，您的评价将有助于我们的系统推荐出更棒的Python代码示例。
示例1: routeListen
def routeListen(self, THRESHOLD=None, MUSIC=False):
"""
Records until a second of silence or times out after 12 seconds
"""
AUDIO_FILE = "location.wav"
RATE = 16000
CHUNK = 1024
LISTEN_TIME = 12
# check if no threshold provided
if THRESHOLD == None:
THRESHOLD = self.fetchThreshold()
self.speaker.play("../static/audio/beep_hi.wav")
# prepare recording stream
audio = pyaudio.PyAudio()
stream = audio.open(format=pyaudio.paInt16,
channels=1,
rate=RATE,
input=True,
frames_per_buffer=CHUNK)
frames = []
# increasing the range # results in longer pause after command generation
lastN = [THRESHOLD * 1.2 for i in range(30)]
for i in range(0, RATE / CHUNK * LISTEN_TIME):
try:
data = stream.read(CHUNK)
except:
pass
frames.append(data)
score = self.getScore(data)
lastN.pop(0)
lastN.append(score)
average = sum(lastN) / float(len(lastN))
# TODO: 0.8 should not be a MAGIC NUMBER!
if average < THRESHOLD * 0.8:
break
self.speaker.play("../static/audio/beep_lo.wav")
# save the audio data
stream.stop_stream()
stream.close()
audio.terminate()
write_frames = open_audio(AUDIO_FILE, 'wb')
write_frames.setnchannels(1)
write_frames.setsampwidth(audio.get_sample_size(pyaudio.paInt16))
write_frames.setframerate(RATE)
write_frames.writeframes(''.join(frames))
write_frames.close()
return self.active_stt_engine.transcribe(AUDIO_FILE, MUSIC=MUSIC, ROUTE=True)
开发者ID:easyNav，项目名称:easyNav-Voice-deprecated-，代码行数:60，代码来源:mic.py
示例2: record
def record(self, time):
audio = PyAudio()
stream = audio.open(input_device_index=self.device_index,
output_device_index=self.device_index,
format=self.format,
channels=self.channel,
rate=self.rate,
input=True,
frames_per_buffer=self.chunk
)
print "Recording..."
frames = []
for i in range(0, self.rate / self.chunk * time):
data = stream.read(self.chunk)
frames.append(data)
stream.stop_stream()
print "Recording Complete"
stream.close()
audio.terminate()
write_frames = open_audio(self.file, 'wb')
write_frames.setnchannels(self.channel)
write_frames.setsampwidth(audio.get_sample_size(self.format))
write_frames.setframerate(self.rate)
write_frames.writeframes(''.join(frames))
write_frames.close()
self.convert()
开发者ID:AnanthaRajuC，项目名称:homecontrol，代码行数:26，代码来源:pygsr.py
示例3: save_audio
def save_audio(self, fp, frames):
sample_width = pyaudio.get_sample_size(self.format)
f = open_audio(fp, 'wb')
f.setsampwidth(sample_width)
f.setframerate(self.rate)
f.setnchannels(1)
f.writeframes(''.join(frames))
f.close()
开发者ID:brunoChr，项目名称:jasper-client，代码行数:8，代码来源:mic.py
示例4: input_speech_passive_listen
def input_speech_passive_listen(self, conf):
"""
Listens and times out after conf['input_speech']['passive_listen_time'].
"""
self._info('Starting Passive Listen')
self.input_speech_get_threshold(conf)
passive_listen_time = conf['input_speech']['passive_listen_time']
rate = conf['input_speech']['rate']
chunk = conf['input_speech']['chunk']
threshold = conf['input_speech']['threshold']
frames = []
sound_heard = False
# start passively listening for disturbance above threshold
for i in range(0, rate / chunk * passive_listen_time):
self.data = self.stream.read(chunk)
frames.append(self.data)
self.input_speech_get_score(conf)
score = conf['input_speech']['score']
if score > threshold:
sound_heard = True
self._info('Heard Something')
# Not sure about invoking mid-sound
#self.invoke('input_speech_passive_sound_heard')
break
if not sound_heard:
self.invoke('input_speech_passive_no_sound_heard')
self._info('No Sound Heard During Passive Listen')
return
# Cut off any recording before the sound was heard
frames = frames[-20:]
# Keep recording for a bit and save the file
delay_multiplier = conf['input_speech']['delay_multiplier']
self._info('Listening...')
for i in range(0, rate / chunk * delay_multiplier):
self.data = self.stream.read(chunk)
frames.append(self.data)
# Save the audio data
self.stream.stop_stream()
self.stream.close()
self.audio.terminate()
write_frames = open_audio(conf['input_speech']['audio_file'], 'wb')
write_frames.setnchannels(1)
write_frames.setsampwidth(self.audio.get_sample_size(pyaudio.paInt16))
write_frames.setframerate(rate)
write_frames.writeframes(''.join(frames))
write_frames.close()
# Transcribe
self.input_speech_transcribe(conf)
开发者ID:edouardpoitras，项目名称:Eva，代码行数:48，代码来源:input_speech.py
示例5: transcribe
def transcribe(self, audio_file_path, PERSONA_ONLY=False, MUSIC=False):
"""
Performs STT via the Google Speech API, transcribing an audio file and returning an English
string.
Arguments:
audio_file_path -- the path to the .wav file to be transcribed
"""
url = "https://www.google.com/speech-api/v2/recognize?output=json&client=chromium&key=%s&lang=%s&maxresults=6&pfilter=2" % (
self.api_key, "pl-PL")
wav = open(audio_file_path, 'rb')
data = wav.read()
wav.close()
f = open_audio(audio_file_path, 'r')
frame_rate = f.getframerate()
f.close()
try:
req = urllib2.Request(
url,
data=data,
headers={
'Content-type': 'audio/l16; rate=%s' % str(frame_rate)})
self.logger.debug("google speech api url: %s frame rate: %d" % (url, frame_rate))
response_url = urllib2.urlopen(req)
response_read = response_url.read()
self.logger.debug("raw response: %s" % repr(response_read))
response_read = response_read.decode('utf-8')
if response_read == '{"result":[]}\n':
text = None
else:
decoded = json.loads(response_read.split("\n")[1])
#self.logger.debug("decoded response: %s" % repr(response_read.decode('utf-8')))
text = decoded['result'][0]['alternative'][0]['transcript']
text = str_formater.unicodeToUTF8(text, self.logger)
if text:
self.logger.info("<<<<<<<<<<<<<<<<<<<")
self.logger.info("YOU: " + text )
self.logger.info("<<<<<<<<<<<<<<<<<<<")
return text
except Exception:
traceback.print_exc()
self.logger.error("Failed to transcribe data: %s" % audio_file_path, exc_info=True)
开发者ID:HubertReX，项目名称:jasper-client，代码行数:44，代码来源:stt.py
示例6: record
def record(self, time=5):
audio = PyAudio()
stream = audio.open(format=self.format, channels=self.channel,
rate=self.rate, input=True,
frames_per_buffer=self.chunk)
print "RECORDING START"
frames = []
for i in range(0, self.rate / self.chunk * time):
data = stream.read(self.chunk)
frames.append(data)
stream.stop_stream()
stream.close()
audio.terminate()
print "RECORDING STOP"
write_frames = open_audio(self.audio_file, 'wb')
write_frames.setnchannels(self.channel)
write_frames.setsampwidth(audio.get_sample_size(self.format))
write_frames.setframerate(self.rate)
write_frames.writeframes(''.join(frames))
write_frames.close()
self.convert()
开发者ID:xSAVIKx，项目名称:py-audio-recognition，代码行数:21，代码来源:recognizer.py
示例7: input_speech_active_listen
def input_speech_active_listen(self, conf):
"""
Records until a conf['input_speech']['listen_time'] amount of seconds
or times out after conf['input_speech']['listen_timeout'] seconds.
"""
rate = conf['input_speech']['rate']
chunk = conf['input_speech']['chunk']
listen_time = conf['input_speech']['listen_time']
threshold = conf['input_speech']['threshold']
self.audio = pyaudio.PyAudio()
self.stream = self.audio.open(format=pyaudio.paInt16,
channels=1,
rate=rate,
input=True,
frames_per_buffer=chunk)
frames = []
lastN = [threshold * 1.2 for i in range(30)]
for i in range(0, rate / chunk * listen_time):
self.data = self.stream.read(chunk)
frames.append(self.data)
self.input_speech_get_score(conf)
score = conf['input_speech']['score']
lastN.pop(0)
lastN.append(score)
average = sum(lastN) / float(len(lastN))
if average < threshold * 0.8: # Why 0.8?
break
# Save the audio data
self.stream.stop_stream()
self.stream.close()
self.audio.terminate()
write_frames = open_audio(conf['input_speech']['audio_file'], 'wb')
write_frames.setnchannels(1)
write_frames.setsampwidth(self.audio.get_sample_size(pyaudio.paInt16))
write_frames.setframerate(rate)
write_frames.writeframes(''.join(frames))
write_frames.close()
# Transcribe
self.input_speech_transcribe(conf)
开发者ID:edouardpoitras，项目名称:Eva，代码行数:39，代码来源:input_speech.py
示例8: record
def record(self, time, device_i=None):
audio = PyAudio()
print audio.get_device_info_by_index(1)
stream = audio.open(input_device_index=device_i,output_device_index=device_i,format=self.format, channels=self.channel,
rate=self.rate, input=True,
frames_per_buffer=self.chunk)
playDing()
print "REC: "
frames = []
for i in range(0, self.rate / self.chunk * time):
data = stream.read(self.chunk)
frames.append(data)
stream.stop_stream()
print "END"
stream.close()
playDing()
audio.terminate()
write_frames = open_audio(self.file, 'wb')
write_frames.setnchannels(self.channel)
write_frames.setsampwidth(audio.get_sample_size(self.format))
write_frames.setframerate(self.rate)
write_frames.writeframes(''.join(frames))
write_frames.close()
开发者ID:sageoffroy，项目名称:Pyranha，代码行数:23，代码来源:vox.py
示例9: passiveListen
def passiveListen(self, PERSONA):
"""
Listens for PERSONA in everyday sound. Times out after LISTEN_TIME, so needs to be
restarted.
"""
THRESHOLD_MULTIPLIER = 1.8
AUDIO_FILE = "passive.wav"
RATE = 16000
CHUNK = 1024
# number of seconds to allow to establish threshold
THRESHOLD_TIME = 1
# number of seconds to listen before forcing restart
LISTEN_TIME = 10
# prepare recording stream
audio = pyaudio.PyAudio()
stream = audio.open(format=pyaudio.paInt16,
channels=1,
rate=RATE,
input=True,
frames_per_buffer=CHUNK)
# stores the audio data
frames = []
# stores the lastN score values
lastN = [i for i in range(30)]
# calculate the long run average, and thereby the proper threshold
for i in range(0, RATE / CHUNK * THRESHOLD_TIME):
data = stream.read(CHUNK)
frames.append(data)
# save this data point as a score
lastN.pop(0)
lastN.append(self.getScore(data))
average = sum(lastN) / len(lastN)
# this will be the benchmark to cause a disturbance over!
THRESHOLD = average * THRESHOLD_MULTIPLIER
# save some memory for sound data
frames = []
# flag raised when sound disturbance detected
didDetect = False
# start passively listening for disturbance above threshold
for i in range(0, RATE / CHUNK * LISTEN_TIME):
data = stream.read(CHUNK)
frames.append(data)
score = self.getScore(data)
if score > THRESHOLD:
didDetect = True
break
# no use continuing if no flag raised
if not didDetect:
print "No disturbance detected"
return
# cutoff any recording before this disturbance was detected
frames = frames[-20:]
# otherwise, let's keep recording for few seconds and save the file
DELAY_MULTIPLIER = 1
for i in range(0, RATE / CHUNK * DELAY_MULTIPLIER):
data = stream.read(CHUNK)
frames.append(data)
# save the audio data
stream.stop_stream()
stream.close()
audio.terminate()
write_frames = open_audio(AUDIO_FILE, 'wb')
write_frames.setnchannels(1)
write_frames.setsampwidth(audio.get_sample_size(pyaudio.paInt16))
write_frames.setframerate(RATE)
write_frames.writeframes(''.join(frames))
write_frames.close()
# check if PERSONA was said
transcribed = self.passive_stt_engine.transcribe(AUDIO_FILE, PERSONA_ONLY=True)
if PERSONA in transcribed:
return (THRESHOLD, PERSONA)
return (False, transcribed)
开发者ID:Glavin001，项目名称:jasper-client，代码行数:95，代码来源:mic.py
示例10: activeListen
def activeListen(self, THRESHOLD=None, LISTEN=True, MUSIC=False):
"""
Records until a second of silence or times out after 12 seconds
"""
AUDIO_FILE = "active.wav"
RATE = 16000
CHUNK = 1024
LISTEN_TIME = 12
# user can request pre-recorded sound
if not LISTEN:
if not os.path.exists(AUDIO_FILE):
return None
return self.transcribe(AUDIO_FILE)
# check if no threshold provided
if THRESHOLD == None:
THRESHOLD = self.fetchThreshold()
os.system("aplay -D hw:1,0 ../static/audio/beep_hi.wav")
# prepare recording stream
audio = pyaudio.PyAudio()
stream = audio.open(format=pyaudio.paInt16,
channels=1,
rate=RATE,
input=True,
frames_per_buffer=CHUNK)
frames = []
# increasing the range # results in longer pause after command
# generation
lastN = [THRESHOLD * 1.2 for i in range(30)]
for i in range(0, RATE / CHUNK * LISTEN_TIME):
data = stream.read(CHUNK)
frames.append(data)
score = self.getScore(data)
lastN.pop(0)
lastN.append(score)
average = sum(lastN) / float(len(lastN))
# TODO: 0.8 should not be a MAGIC NUMBER!
if average < THRESHOLD * 0.8:
break
os.system("aplay -D hw:1,0 ../static/audio/beep_lo.wav")
# save the audio data
stream.stop_stream()
stream.close()
audio.terminate()
write_frames = open_audio(AUDIO_FILE, 'wb')
write_frames.setnchannels(1)
write_frames.setsampwidth(audio.get_sample_size(pyaudio.paInt16))
write_frames.setframerate(RATE)
write_frames.writeframes(''.join(frames))
write_frames.close()
# DO SOME AMPLIFICATION
# os.system("sox "+AUDIO_FILE+" temp.wav vol 20dB")
if MUSIC:
return self.transcribe(AUDIO_FILE, MUSIC=True)
return self.transcribe(AUDIO_FILE)
开发者ID:Asalion，项目名称:jasper-client，代码行数:71，代码来源:mic.py
示例11: loadAudio
def loadAudio(self, AUDIO_FILE, THRESHOLD=None, LISTEN=True, MUSIC=False, RATE=48000, CHUNK=8096, LISTEN_TIME=5, AVERAGE_TIME=None):
"""
Records until a second of silence or times out after 12 seconds
"""
#AUDIO_FILE = "active.wav"
if not AUDIO_FILE:
print "No WAV file name given"
return None
else:
if not os.path.isfile(AUDIO_FILE):
print "Given WAV faile doesn't exist: %s " % AUDIO_FILE
return None
read_frames = open_audio(AUDIO_FILE, 'rb')
self.CHANNELS = read_frames.getnchannels()
sample_size = read_frames.getsampwidth()
#self.audio.get_sample_size(self.FORMAT)
self.RATE = read_frames.getframerate()
#self.RATE = RATE
self.CHUNK = CHUNK
THRESHOLD_LIMIT_RATIO = 1.0 #0.8
#RATE = 16000
#LISTEN_TIME = 5
if not AVERAGE_TIME:
AVERAGE_TIME = LISTEN_TIME
LAST_SAMPLES_NO = int(AVERAGE_TIME * (self.RATE / self.CHUNK))
#LAST_SAMPLES_NO = 5
# check if no threshold provided
if THRESHOLD == None:
if not self.THRESHOLD:
self.THRESHOLD = self.fetchThreshold(RATE=RATE, CHUNK=CHUNK)
else:
self.THRESHOLD = THRESHOLD
self.logger.debug("THRESHOLD: %6.2f" % self.THRESHOLD)
limit = round(self.THRESHOLD * THRESHOLD_LIMIT_RATIO * 100.0) / 100.0
#self.speaker.play("../static/audio/beep_hi.mp3")
# wait 330 ms in order not to record beep
#time.sleep(0.33)
# prepare recording stream
#audio = pyaudio.PyAudio()
#defaultSampleRate = audio.get_device_info_by_index(0)['defaultSampleRate']
#self.logger.debug("defaultSampleRate: %s" % repr(defaultSampleRate))
frames = []
# increasing the range # results in longer pause after command generation
#lastN = [THRESHOLD * 1.2 for i in range(30)]
lastN = []
allN = []
#self.logger.debug("lastN: %s" % repr(lastN))
while True:
try:
data = read_frames.readframes(self.CHUNK)
if not data:
print "got end of file"
break
#data = self.audioFilter(data)
frames.append(data)
score = self.getScore(data)
score = self.to_dB(score)
#score = self.get_RMS(data)
#score = self.get_dB(score)
score = round(score * 100.0) / 100.0
if len(lastN) >= LAST_SAMPLES_NO:
lastN.pop(0)
lastN.append(score)
allN.append(score)
#self.logger.debug("lastN: %s" % repr(lastN))
average = sum(lastN) / float(len(lastN))
#self.logger.debug("score: %6.2f average: %6.2f THRESHOLD : %6.2f" % (score, average, THRESHOLD))
cut_off = 120.0
#cut_off = 999.0
split = LAST_SAMPLES_NO
header = '[ Current: %10.2f | Average: %10.2f | Threshold : %10.2f | Cut off: %10.2f | Average time: %4d s ]\n' % (score, average, limit, cut_off, AVERAGE_TIME)
self.volume_bar.draw_bar(allN, header, limit, self.RATE / self.CHUNK, cut_off, split)
arr = self.extractData(data)
print len(arr), min(arr), max(arr), sum(arr)/len(arr)
text = raw_input("<pause>")
except IOError:
self.logger.critical("IOError error reading chunk", exc_info=True)
break
except KeyboardInterrupt:
print 'got break'
#.........这里部分代码省略.........
开发者ID:HubertReX，项目名称:jasper-client，代码行数:101，代码来源:half_local_mic.py
示例12: recordAudio
#.........这里部分代码省略.........
self.THRESHOLD = THRESHOLD
self.THRESHOLD = abs(self.THRESHOLD)
self.logger.debug("THRESHOLD: %6.2f" % self.THRESHOLD)
limit = round(self.THRESHOLD * THRESHOLD_LIMIT_RATIO * 100.0) / 100.0
#self.speaker.play("../static/audio/beep_hi.mp3")
# wait 330 ms in order not to record beep
#time.sleep(0.33)
# prepare recording stream
#audio = pyaudio.PyAudio()
#defaultSampleRate = audio.get_device_info_by_index(0)['defaultSampleRate']
#self.logger.debug("defaultSampleRate: %s" % repr(defaultSampleRate))
stream = self.audio.open(format = self.FORMAT,
channels = self.CHANNELS,
input_device_index = self.INPUT_DEVICE_IDX,
rate = self.RATE,
input = True,
frames_per_buffer = self.CHUNK)
frames = []
# increasing the range # results in longer pause after command generation
#lastN = [THRESHOLD * 1.2 for i in range(30)]
lastN = []
allN = []
#self.logger.debug("lastN: %s" % repr(lastN))
wasAbove = False
wasBelow = False
gotKeyboardInterrupt = False
#self.volume_bar.save()
for i in range(0, self.RATE / self.CHUNK * LISTEN_TIME):
try:
data = stream.read(self.CHUNK)
#data = self.audioFilter(data)
if len(frames) >= LAST_FRAMES_NO:
frames.pop(0)
frames.append(data)
#score = round(self.getScore(data) * 100.0) / 100.0
score = round(self.to_dB(self.getScore(data)) * 100.0) / 100.0
if len(lastN) >= LAST_SAMPLES_NO:
lastN.pop(0)
lastN.append(score)
allN.append(score)
#self.logger.debug("lastN: %s" % repr(lastN))
average = sum(lastN) / float(len(lastN))
#self.logger.debug("score: %6.2f average: %6.2f THRESHOLD : %6.2f" % (score, average, THRESHOLD ))
cut_off = 120.0
split = LAST_SAMPLES_NO
header = '[ Current: %10.2f | Average: %10.2f | Threshold : %10.2f | Cut off: %10.2f | Average time: %4d s | was Above: %d ]\n' % (score, average, limit, cut_off, AVERAGE_TIME, wasAbove)
self.volume_bar.draw_bar(allN, header, limit, self.RATE / self.CHUNK, cut_off, split, verbose=True)
if not wasAbove and self.isAboveThreshold(lastN, limit):
wasAbove = True
if wasAbove and self.isBelowThreshold(lastN, limit):
print "not above threshold any more"
wasBelow = True
break
#if average < limit and len(lastN) == LAST_SAMPLES_NO:
# break
except IOError:
self.logger.critical("IOError error reading chunk", exc_info=True)
except KeyboardInterrupt:
print 'got break'
# temporarly mask exception to clean up
gotKeyboardInterrupt = True
break
#self.speaker.play("../static/audio/beep_lo.mp3")
# save the audio data
stream.stop_stream()
stream.close()
#self.audio.terminate()
if wasBelow:
write_frames = open_audio(AUDIO_FILE, 'wb')
write_frames.setnchannels(self.CHANNELS)
write_frames.setsampwidth(self.audio.get_sample_size(self.FORMAT))
write_frames.setframerate(self.RATE)
write_frames.writeframes(''.join(frames))
write_frames.close()
else:
#finished after timeout and not threshold crossed - not record audio to file
AUDIO_FILE = None
#self.volume_bar.restore()
if gotKeyboardInterrupt:
# all is cleaned up - rerise exception
#raise KeyboardInterrupt
return None
return AUDIO_FILE
开发者ID:HubertReX，项目名称:jasper-client，代码行数:101，代码来源:half_local_mic.py
示例13: activeListen
def activeListen(self, THRESHOLD=None, RATE=48000, CHUNK=8096):
"""
Records until a second of silence or times out after 12 seconds
"""
AUDIO_FILE = "active.wav"
self.RATE = RATE
self.CHUNK = CHUNK
#RATE = 16000
#RATE = 44100
#CHUNK = 512
LISTEN_TIME = 5
LAST_SAMPLES_NO = 10
# check if no threshold provided
if THRESHOLD == None:
THRESHOLD = self.fetchThreshold()
self.logger.debug("THRESHOLD: %6.2f" % THRESHOLD)
self.play("../static/audio/beep_hi.mp3")
# prepare recording stream
#audio = pyaudio.PyAudio()
#defaultSampleRate = self.audio.get_device_info_by_index(0)['defaultSampleRate']
#self.logger.debug("defaultSampleRate: %s" % repr(defaultSampleRate))
stream = self.audio.open(format=self.FORMAT,
channels=self.CHANNELS,
input_device_index=0,
rate=self.RATE,
input=True,
frames_per_buffer=self.CHUNK)
frames = []
# increasing the range # results in longer pause after command generation
#lastN = [THRESHOLD * 1.2 for i in range(30)]
lastN = []
#self.logger.debug("lastN: %s" % repr(lastN))
for i in range(0, self.RATE / self.CHUNK * LISTEN_TIME):
try:
data = stream.read(self.CHUNK)
frames.append(data)
score = round(self.getScore(data) * 100.0) / 100.0
if len(lastN) >= LAST_SAMPLES_NO:
lastN.pop(0)
lastN.append(score)
#self.logger.debug("lastN: %s" % repr(lastN))
average = sum(lastN) / float(len(lastN))
self.logger.debug("score: %6.2f average: %6.2f THRESHOLD * 0.8: %6.2f" % (score, average, THRESHOLD * 0.8))
# TODO: 0.8 should not be a MAGIC NUMBER!
if average < THRESHOLD * 0.8 and len(lastN) == LAST_SAMPLES_NO:
break
except IOError:
self.logger.critical("IOError error reading chunk", exc_info=True)
self.play("../static/audio/beep_lo.mp3")
# save the audio data
stream.stop_stream()
stream.close()
#self.audio.terminate()
write_frames = open_audio(AUDIO_FILE, 'wb')
write_frames.setnchannels(self.CHANNELS)
write_frames.setsampwidth(self.audio.get_sample_size(self.FORMAT))
write_frames.setframerate(self.RATE)
write_frames.writeframes(''.join(frames))
write_frames.close()
return AUDIO_FILE
开发者ID:HubertReX，项目名称:jasper-client，代码行数:73，代码来源:setup_mic.py
示例14: activeListen
def activeListen(self, THRESHOLD=None, LISTEN=True, MUSIC=False):
"""
Records until a second of silence or times out after 12 seconds
"""
AUDIO_FILE = "active.wav"
RATE = 16000
#RATE = 44100
CHUNK = 1024
LISTEN_TIME = 12
# user can request pre-recorded sound
if not LISTEN:
if not os.path.exists(AUDIO_FILE):
return None
return self.active_stt_engine.transcribe(AUDIO_FILE)
# check if no threshold provided
if THRESHOLD == None:
THRESHOLD = self.fetchThreshold()
self.speaker.play("../static/audio/beep_hi.mp3")
# prepare recording stream
audio = pyaudio.PyAudio()
defaultSampleRate = audio.get_device_info_by_index(0)['defaultSampleRate']
self.logger.debug("defaultSampleRate: %s" % repr(defaultSampleRate))
stream = audio.open(format=pyaudio.paInt16,
channels=1,
rate=RATE,
input=True,
frames_per_buffer=CHUNK)
frames = []
# increasing the range # results in longer pause after command generation
lastN = [THRESHOLD * 1.2 for i in range(30)]
for i in range(0, RATE / CHUNK * LISTEN_TIME):
try:
data = stream.read(CHUNK)
frames.append(data)
score = self.getScore(data)
lastN.pop(0)
lastN.append(score)
average = sum(lastN) / float(len(lastN))
# TODO: 0.8 should not be a MAGIC NUMBER!
if average < THRESHOLD * 0.8:
break
except IOError:
self.logger.critical("IOError error reading chunk", exc_info=True)
self.speaker.play("../static/audio/beep_lo.mp3")
# save the audio data
stream.stop_stream()
stream.close()
audio.terminate()
write_frames = open_audio(AUDIO_FILE, 'wb')
write_frames.setnchannels(1)
write_frames.setsampwidth(audio.get_sample_size(pyaudio.paInt16))
write_frames.setframerate(RATE)
write_frames.writeframes(''.join(frames))
write_frames.close()
return self.active_stt_engine.transcribe(AUDIO_FILE, MUSIC)
开发者ID:HubertReX，项目名称:jasper-client，代码行数:70，代码来源:mic.py
示例15: listen
def listen(self, level = 1000,timeout = 1,ignore_shoter_than = 0.5,ignore_longer_than = 5 ,language = "sv_SE", device_i=None):
audio = PyAudio()
#print audio.get_device_info_by_index(1)
stream = audio.open(input_device_index=device_i,output_device_index=device_i,format=self.format, channels=self.channel,
rate=self.rate, input=True,
frames_per_buffer=self.chunk)
timeout_chuncks = self.rate / self.chunk * timeout
minmessage = self.rate / self.chunk * ignore_shoter_than
maxmessage = self.rate / self.chunk * ignore_longer_than
try:
while(True):
print "Start listening... "
frames = []
data = ""
olddata = ""
self.count_silence = 0
self.active = False
while(True): #for i in range(0, self.rate / self.chunk * time):
data = stream.read(self.chunk)
rms = audioop.rms(data, 2)
#print str(rms) + '\r'
#There is some noise start recording
if rms > level:
self.count_silence = 0
if self.active == False:
print "Recording..."
self.active = True
self.count_silence = 0
frames.append(olddata)
if self.active:
frames.append(data)
if rms < level and self.active:
self.count_silence += 1
#If we have enough silence send for processing
if (self.count_silence > timeout_chuncks) and self.active == True:
self.active = False
#print len(frames) #10 12
#print self.count_silence #8
if not len(frames)> self.count_silence + minmessage:
print "Disregarding noise"
frames = []
continue
if len(frames)> self.count_silence + maxmessage:
print "Ignoring to long recording"
frames = []
continue
print "Processing..."
break
olddata = data
write_frames = open_audio(self.file, 'wb')
write_frames.setnchannels(self.channel)
write_frames.setsampwidth(audio.get_sample_size(self.format))
write_frames.setframerate(self.rate)
write_frames.writeframes(''.join(frames))
write_frames.close()
self.convert()
try:
phrase, complete_response = self.speech_to_text(language) # select the language
except:
phrase = ""
print phrase
except KeyboardInterrupt:
# quit
stream.stop_stream()
#print "END"
stream.close()
audio.terminate()
sys.exit()
return
开发者ID:Anton04，项目名称:Speech2MQTT，代码行数:87，代码来源:Speech2MQTT.py
注：本文中的wave.open_audio函数示例由纯净天空整理自Github/MSDocs等源码及文档管理平台，相关代码片段筛选自各路编程大神贡献的开源项目，源码版权归原作者所有，传播和使用请参考对应项目的License；未经允许，请勿转载。 |
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