Python alsaaudio.PCM属性代码示例

# 需要导入模块: import alsaaudio [as 别名]
# 或者: from alsaaudio import PCM [as 别名]
def start():
	last = GPIO.input(button)
	while True:
		val = GPIO.input(button)
		GPIO.wait_for_edge(button, GPIO.FALLING) # we wait for the button to be pressed
		GPIO.output(lights[1], GPIO.HIGH)
		inp = alsaaudio.PCM(alsaaudio.PCM_CAPTURE, alsaaudio.PCM_NORMAL, device)
		inp.setchannels(1)
		inp.setrate(16000)
		inp.setformat(alsaaudio.PCM_FORMAT_S16_LE)
		inp.setperiodsize(500)
		audio = ""
		while(GPIO.input(button)==0): # we keep recording while the button is pressed
			l, data = inp.read()
			if l:
				audio += data
		rf = open(path+'recording.wav', 'w')
		rf.write(audio)
		rf.close()
		inp = None
		alexa() 

# 需要导入模块: import alsaaudio [as 别名]
# 或者: from alsaaudio import PCM [as 别名]
def play(f):    
    device = alsaaudio.PCM()
    device.setchannels(f.getnchannels())
    device.setrate(f.getframerate())
    if f.getsampwidth() == 1:
        device.setformat(alsaaudio.PCM_FORMAT_U8)
    # Otherwise we assume signed data, little endian
    elif f.getsampwidth() == 2:
        device.setformat(alsaaudio.PCM_FORMAT_S16_LE)
    elif f.getsampwidth() == 3:
        device.setformat(alsaaudio.PCM_FORMAT_S24_LE)
    elif f.getsampwidth() == 4:
        device.setformat(alsaaudio.PCM_FORMAT_S32_LE)
    else:
        raise ValueError('Unsupported format')
    device.setperiodsize(320)
    data = f.readframes(320)
    while data:
        # Read data from stdin
        device.write(data)
        data = f.readframes(320) 

# 需要导入模块: import alsaaudio [as 别名]
# 或者: from alsaaudio import PCM [as 别名]
def doring(self):
        if self.ringfile is not None:
            self.ringfile.rewind()
        else:
            self.ringfile = wave.open(self.config["soundfiles"]["ringtone"], 'rb')
            self.device = alsaaudio.PCM(card="pulse")
            self.device.setchannels(self.ringfile.getnchannels())
            self.device.setrate(self.ringfile.getframerate())
            self.device.setperiodsize(320)


        while self.shouldring:
            data = self.ringfile.readframes(320)
            while data:
                self.device.write(data)
                data = self.ringfile.readframes(320)

            self.ringfile.rewind()
            time.sleep(2)
            if time.time() - 60 > self.ringstart:
                self.stop() 

# 需要导入模块: import alsaaudio [as 别名]
# 或者: from alsaaudio import PCM [as 别名]
def run(self):
        # open audio file and device
        audio_file = wave.open(self.soundfile, 'rb')
        audio_device = alsaaudio.PCM(alsaaudio.PCM_PLAYBACK, alsaaudio.PCM_NORMAL, 'default')

        # we are hard coding the audio format!
        audio_device.setchannels(2)
        audio_device.setrate(44100)
        audio_device.setformat(alsaaudio.PCM_FORMAT_S16_LE)
        audio_device.setperiodsize(980)

        # play the audio
        audio_data = audio_file.readframes(980)
        while audio_data:
          audio_device.write(audio_data)
          audio_data = audio_file.readframes(980)

        audio_file.close()

# Class for blinking the leds 

# 需要导入模块: import alsaaudio [as 别名]
# 或者: from alsaaudio import PCM [as 别名]
def acquire(self):
        if self._session.is_active():
            try:
                pcm_args = {
                    'type': alsa.PCM_PLAYBACK,
                    'mode': alsa.PCM_NORMAL,
                }
                if self._args.playback_device != 'default':
                    pcm_args['device'] = self._args.playback_device
                else:
                    pcm_args['card'] = self._args.device
                pcm = alsa.PCM(**pcm_args)

                pcm.setchannels(CHANNELS)
                pcm.setrate(RATE)
                pcm.setperiodsize(PERIODSIZE)
                pcm.setformat(alsa.PCM_FORMAT_S16_LE)

                self._device = pcm
                print "AlsaSink: device acquired"
            except alsa.ALSAAudioError as error:
                print "Unable to acquire device: ", error
                self.release() 

# 需要导入模块: import alsaaudio [as 别名]
# 或者: from alsaaudio import PCM [as 别名]
def open_stream(self, bits, channels, rate, chunksize=1024, output=True):
        # Check if format is supported
        is_supported_fmt = self.supports_format(bits, channels, rate,
                                                output=output)
        if not is_supported_fmt:
            msg_fmt = ("ALSAAudioDevice ({name}) doesn't support " +
                       "%s format (Int{bits}, {channels}-channel at" +
                       " {rate} Hz)") % ('output' if output else 'input')
            msg = msg_fmt.format(name=self.name,
                                 bits=bits,
                                 channels=channels,
                                 rate=rate)
            self._logger.critical(msg)
            raise Exception(msg)
        # Everything looks fine, open the PCM stream
        pcm_type = alsaaudio.PCM_PLAYBACK if output else alsaaudio.PCM_CAPTURE
        stream = alsaaudio.PCM(type=pcm_type,
                               mode=alsaaudio.PCM_NORMAL,
                               device=self.name)
        stream.setchannels(channels)
        stream.setrate(rate)
        stream.setformat(bits_to_samplefmt(bits))
        stream.setperiodsize(chunksize)
        self._logger.debug("%s stream opened on device '%s' (%d Hz, %d " +
                           "channel, %d bit)", "output" if output else "input",
                           self.slug, rate, channels, bits)
        try:
            yield stream
        finally:
            stream.close()
            self._logger.debug("%s stream closed on device '%s'",
                               "output" if output else "input", self.slug) 

# 需要导入模块: import alsaaudio [as 别名]
# 或者: from alsaaudio import PCM [as 别名]
def playhandset(self):
        print "Starting dialtone"
        wv = wave.open(self.handsetfile)
        device = alsaaudio.PCM(card="plug:external")
        #device.setchannels(wv.getnchannels())
        #device.setrate(wv.getframerate())
        #device.setperiodsize(320)

        data = wv.readframes(320)
        while data and self.shouldplayhandset:
            device.write(data)
            data = wv.readframes(320)
        wv.rewind()
        wv.close() 

# 需要导入模块: import alsaaudio [as 别名]
# 或者: from alsaaudio import PCM [as 别名]
def playfile(self, file):
        wv = wave.open(file)
        self.device = alsaaudio.PCM(card="pulse")
        self.device.setchannels(wv.getnchannels())
        self.device.setrate(wv.getframerate())
        self.device.setperiodsize(320)

        data = wv.readframes(320)
        while data:
            self.device.write(data)
            data = wv.readframes(320)
        wv.rewind()
        wv.close() 

# 需要导入模块: import alsaaudio [as 别名]
# 或者: from alsaaudio import PCM [as 别名]
def play(self, file_path):

        if self.convert:
            self.convert_mp3_to_wav(file_path_mp3=file_path)
        f = wave.open(file_path, 'rb')
        pcm_type = alsaaudio.PCM_PLAYBACK
        stream = alsaaudio.PCM(type=pcm_type,
                               mode=alsaaudio.PCM_NORMAL,
                               device=self.device)
        # Set attributes
        stream.setchannels(f.getnchannels())
        stream.setrate(f.getframerate())
        bits = f.getsampwidth()*8
        stream.setformat(bits_to_samplefmt(bits))        
        stream.setperiodsize(CHUNK)
        
        logger.debug("[PyAlsaAudioPlayer] %d channels, %d sampling rate, %d bit" % (f.getnchannels(),
                                                                                    f.getframerate(),
                                                                                    bits))
        
        data = f.readframes(CHUNK)
        while data:
            # Read data from stdin
            stream.write(data)
            data = f.readframes(CHUNK)
     
        f.close()
        stream.close() 

# 需要导入模块: import alsaaudio [as 别名]
# 或者: from alsaaudio import PCM [as 别名]
def open_stream(self, bits, channels, rate, chunksize=1024, output=True):
        # Check if format is supported
        is_supported_fmt = self.supports_format(bits, channels, rate,
                                                output=output)
        if not is_supported_fmt:
            msg_fmt = ("ALSAAudioDevice ({name}) doesn't support " +
                       "%s format (Int{bits}, {channels}-channel at" +
                       " {rate} Hz)") % ('output' if output else 'input')
            msg = msg_fmt.format(name=self.name,
                                 bits=bits,
                                 channels=channels,
                                 rate=rate)
            self._logger.critical(msg)
            raise Exception(msg)
        # Everything looks fine, open the PCM stream
        pcm_type = alsaaudio.PCM_PLAYBACK if output else alsaaudio.PCM_CAPTURE
        stream = alsaaudio.PCM(type=pcm_type,
                               mode=alsaaudio.PCM_NORMAL,
                               device='default')
        stream.setchannels(channels)
        stream.setrate(rate)
        stream.setformat(bits_to_samplefmt(bits))
        stream.setperiodsize(chunksize)
        self._logger.debug("%s stream opened on device '%s' (%d Hz, %d " +
                           "channel, %d bit)", "output" if output else "input",
                           self.slug, rate, channels, bits)
        try:
            yield stream
        finally:
            stream.close()
            self._logger.debug("%s stream closed on device '%s'",
                               "output" if output else "input", self.slug) 

# 需要导入模块: import alsaaudio [as 别名]
# 或者: from alsaaudio import PCM [as 别名]
def _record(self, cb):
        inp = None
        try:
            inp = alsaaudio.PCM(
                alsaaudio.PCM_CAPTURE,
                alsaaudio.PCM_NORMAL,
                device=self.record_device,
            )
            inp.setchannels(1)
            inp.setrate(16000)
            inp.setformat(alsaaudio.PCM_FORMAT_S16_LE)
            inp.setperiodsize(1600)  # 100ms
            finalize = False
            while not finalize:
                l, data = inp.read()
                if not self.currently_recording:
                    finalize = True
                if l or finalize:
                    # self.recorded_raw.write(data)
                    cb(data, finalize)
        except Exception:
            print(traceback.format_exc())
        finally:
            self.currently_recording = False
            if inp:
                inp.close() 

# 需要导入模块: import alsaaudio [as 别名]
# 或者: from alsaaudio import PCM [as 别名]
def recordAudioSegments(RecordPath, BLOCKSIZE):	
	# This function is used for recording audio segments (until ctr+c is pressed)
	# ARGUMENTS:
	# - RecordPath:		the path where the wav segments will be stored
	# - BLOCKSIZE:		segment recording size (in seconds)
	# 
	# NOTE: filenames are based on clock() value
	
	print "Press Ctr+C to stop recording"
	RecordPath += os.sep
	d = os.path.dirname(RecordPath)
	if os.path.exists(d) and RecordPath!=".":
		shutil.rmtree(RecordPath)	
	os.makedirs(RecordPath)	

	inp = alsaaudio.PCM(alsaaudio.PCM_CAPTURE,alsaaudio.PCM_NONBLOCK)
	inp.setchannels(1)
	inp.setrate(Fs)
	inp.setformat(alsaaudio.PCM_FORMAT_S16_LE)
	inp.setperiodsize(512)
	midTermBufferSize = int(Fs*BLOCKSIZE)
	midTermBuffer = []
	curWindow = []
	elapsedTime = "%08.3f" % (time.time())
	while 1:
			l,data = inp.read()		   
		    	if l:
				for i in range(len(data)/2):
					curWindow.append(audioop.getsample(data, 2, i))
		
				if (len(curWindow)+len(midTermBuffer)>midTermBufferSize):
					samplesToCopyToMidBuffer = midTermBufferSize - len(midTermBuffer)
				else:
					samplesToCopyToMidBuffer = len(curWindow)

				midTermBuffer = midTermBuffer + curWindow[0:samplesToCopyToMidBuffer];
				del(curWindow[0:samplesToCopyToMidBuffer])
			

			if len(midTermBuffer) == midTermBufferSize:
				# allData = allData + midTermBuffer				
				curWavFileName = RecordPath + os.sep + str(elapsedTime) + ".wav"				
				midTermBufferArray = numpy.int16(midTermBuffer)
				wavfile.write(curWavFileName, Fs, midTermBufferArray)
				print "AUDIO  OUTPUT: Saved " + curWavFileName
				midTermBuffer = []
				elapsedTime = "%08.3f" % (time.time()) 

# 需要导入模块: import alsaaudio [as 别名]
# 或者: from alsaaudio import PCM [as 别名]
def __test_device(device, record):
        """
            Test selected ALSA device, making sure it handles both stereo and
            mono and both 44.1KHz and 22.05KHz on output, mono and 16 kHz on
            input.

            On a typical RPI configuration, default with hifiberry card is not
            configured to do software-mono, so we'll use
            plughw:CARD=sndrpihifiberry instead.
            Likewise, on 2019 cards, hw:CARD=seeed2micvoicec is not able to run
            mono sound.

            @param device: name of the sound device
            @type device: six.text_type
            @param record: C{True} if this method is looking for recording
            device. C{False} if the device should only playback.
            @type record: bool
        """
        try:
            dev = None

            if record:
                dev = alsaaudio.PCM(alsaaudio.PCM_CAPTURE, device=device)
            else:
                dev = alsaaudio.PCM(device=device)

            if (
                dev.setformat(alsaaudio.PCM_FORMAT_S16_LE)
                != alsaaudio.PCM_FORMAT_S16_LE
            ):
                return False
            if record:
                if dev.setchannels(1) != 1:
                    return False
                if dev.setrate(16000) != 16000:
                    return False
            else:
                if dev.setchannels(2) != 2:
                    return False
                if dev.setchannels(1) != 1:
                    return False
                if dev.setrate(44100) != 44100:
                    return False
                if dev.setrate(22050) != 22050:
                    return False
        except alsaaudio.ALSAAudioError:
            return False
        finally:
            if dev:
                dev.close()
        return True