'''
The MIT License (MIT)
Copyright (c) 2017 Sean UN Wood
Permission is hereby granted, free of charge, to any person obtaining a copy
of this software and associated documentation files (the "Software"), to deal
in the Software without restriction, including without limitation the rights
to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
copies of the Software, and to permit persons to whom the Software is
furnished to do so, subject to the following conditions:
The above copyright notice and this permission notice shall be included in all
copies or substantial portions of the Software.
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
SOFTWARE.
@author: Sean UN Wood
'''
import logging
import numpy as np
from multiprocessing import Process
from time import sleep
import time as tm
from gccNMF.wavfile import pcm2float, float2pcm
class PyAudioStreamProcessor(Process):
def __init__(self, numChannels, sampleRate, windowSize, hopSize, blockSize, deviceIndex,
togglePlayQueue, togglePlayAck, inputFrames, outputFrames, processFramesEvent, processFramesDoneEvent, terminateEvent):
super(PyAudioStreamProcessor, self).__init__()
self.numChannels = numChannels
self.sampleRate = sampleRate
self.windowSize = windowSize
self.hopSize = hopSize
self.blockSize = blockSize
self.togglePlayQueue = togglePlayQueue
self.togglePlayAck = togglePlayAck
self.inputFrames = inputFrames
self.outputFrames = outputFrames
self.processFramesEvent = processFramesEvent
self.processFramesDoneEvent = processFramesDoneEvent
self.terminateEvent = terminateEvent
self.deviceIndex = deviceIndex
self.numBlocksPerBuffer = 8
self.processingTimes = []
self.underflowCounter = 0
self.fileName = None
self.audioStream = None
self.pyaudio = None
self.fileNameChanged = False
def run(self):
#os.nice(-20)
lastPrintTime = tm.time()
while True:
currentTime = tm.time()
if self.terminateEvent.is_set():
logging.debug('AudioStreamProcessor: received terminate')
if self.audioStream:
self.audioStream.close()
logging.debug('AudioStreamProcessor: stream stopped')
return
if not self.togglePlayQueue.empty():
parameters = self.togglePlayQueue.get()
logging.debug('AudioStreamProcessor: received togglePlayParams')
fileName = parameters['fileName']
if fileName != self.fileName:
self.fileName = fileName
self.fileNameChanged = True
if self.active():
self.reset()
if 'stop' in parameters: self.stopStream()
elif 'start' in parameters: self.startStream()
logging.debug('AudioStreamProcessor: processed togglePlayParams')
self.togglePlayAck.set()
logging.debug('AudioStreamProcessor: ack set')
elif currentTime - lastPrintTime >= 2:
if len(self.processingTimes) != 0:
logging.info( 'Processing times (min/max/avg): %f, %f, %f' % (np.min(self.processingTimes), np.max(self.processingTimes), np.mean(self.processingTimes)) )
lastPrintTime = currentTime
del self.processingTimes[:]
else:
sleep(0.1)
def filePlayerCallback(self, in_data, numFrames, time_info, status):
startTime = tm.time()
if self.sampleIndex+numFrames >= self.numFrames:
self.sampleIndex = 0
inputBuffer = self.samples[self.sampleIndex*self.bytesPerFrameAllChannels:(self.sampleIndex+numFrames)*self.bytesPerFrameAllChannels]
inputIntArray = np.frombuffer(inputBuffer, dtype='<i2')
self.inputFrames[:] = pcm2float(inputIntArray).reshape(-1, self.numChannels).T
self.sampleIndex += numFrames
#logging.info('AudioStreamProcessor: setting processFramesEvent')
self.processFramesDoneEvent.clear()
self.processFramesEvent.set()
#logging.info('AudioStreamProcessor: waiting for processFramesDoneEvent')
self.processFramesDoneEvent.wait()
#logging.info('AudioStreamProcessor: done waiting for processFramesDoneEvent')
outputIntArray = float2pcm(self.outputFrames.T.flatten())
try:
outputBuffer = np.getbuffer(outputIntArray)
except:
outputBuffer = outputIntArray.tobytes()
self.processingTimes.append(tm.time() - startTime)
return outputBuffer, self.paContinue
def active(self):
if not self.audioStream:
return False
else:
return self.audioStream.is_active()
def startStream(self):
if not self.audioStream or self.fileNameChanged:
logging.info('AudioStreamProcessor: creating stream...')
self.fileNameChanged = False
self.reset()
logging.info('AudioStreamProcessor: starting stream')
self.audioStream.start_stream()
def stopStream(self):
if self.audioStream:
logging.info('AudioStreamProcessor: stopping stream')
self.audioStream.stop_stream()
def reset(self):
if self.audioStream:
logging.info('AudioStreamProcessor: aborting stream')
self.audioStream.close()
self.createAudioStream()
def togglePlay(self):
self.stopStream() if self.active() else self.startStream()
def logProcessingTimes(self):
if len(self.processingTimes) == 0:
return
with self.processingTimesLock:
minProcessingTime = np.min(self.processingTimes)
maxProcessingTime = np.max(self.processingTimes)
meanProcessingTime = np.mean(self.processingTimes)
stdProcessingTime = np.std(self.processingTimes)
minTimeToProcess = np.min(self.timesToProcess)
maxTimeToProcess = np.max(self.timesToProcess)
meanTimeToProcess = np.mean(self.timesToProcess)
stdTimeToProcess = np.std(self.timesToProcess)
del self.processingTimes[:]
with self.underflowCounterLock:
numUnderflows = self.underflowCounter
self.underflowCounter = 0
logging.info( 'Min/max/mean/std processing time: %f, %f, %f, %f. Num underflows: %d (min/max/meanTimeToProcess' % (minProcessingTime, maxProcessingTime, meanProcessingTime, stdProcessingTime, numUnderflows) )
logging.info( 'min/max/mean/std time to process: %f, %f, %f, %f' % (minTimeToProcess, maxTimeToProcess, meanTimeToProcess, stdTimeToProcess) )
def createAudioStream(self):
import wave
import pyaudio
if self.pyaudio is None:
self.pyaudio = pyaudio.PyAudio()
self.paContinue = pyaudio.paContinue
waveFile = wave.open(self.fileName, 'rb')
self.numFrames = waveFile.getnframes()
self.samples = waveFile.readframes(self.numFrames)
self.sampleRate = waveFile.getframerate()
self.bytesPerFrame = waveFile.getsampwidth()
self.bytesPerFrameAllChannels = self.bytesPerFrame * self.numChannels
self.format = self.pyaudio.get_format_from_width(self.bytesPerFrame)
waveFile.close()
self.waveFile = wave.open(self.fileName, 'rb')
self.sampleIndex = 0
self.audioStream = self.pyaudio.open(format=self.format,
channels=self.numChannels,
rate=self.sampleRate,
frames_per_buffer=self.blockSize,
output=True,
stream_callback=self.filePlayerCallback)
