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PyAudio
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#! /bin/bash
sudo apt-get update
sudo apt-get install python
sudo apt-get install python-pyaudio
sudo apt-get install python-numpy
sudo apt-get install python-pyplot
sudo apt-get install python-matplotlib
sudo apt-get install python-scipy
sudo apt-get autoremove
exit#! /usr/bin/python2.7
import pyaudio
def DevsInfo():
p=pyaudio.PyAudio()
print "\nSoundCards available"
for i in range(p.get_default_host_api_info()["deviceCount"]):
if p.get_device_info_by_index(i)["maxInputChannels"]>0:
DevId = p.get_device_info_by_index(i)["index"]
DevName = p.get_device_info_by_index(i)["name"]
DevInCh = p.get_device_info_by_index(i)["maxInputChannels"]
DevOutCh = p.get_device_info_by_index(i)["maxOutputChannels"]
DevSRate = p.get_device_info_by_index(i)["defaultSampleRate"]
print "Id=%d\tName=%s\tMaxInChannels=%d\tMaxOutChannels=%d\tDefaultSampleRate=%d"%(DevId,DevName,DevInCh,DevOutCh,DevSRate)
DevsInfo()$ ./sho_soundcards.py
ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear
ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe
ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side
ALSA lib pcm_dmix.c:961:(snd_pcm_dmix_open) The dmix plugin supports only playback stream
SoundCards available
Id=0 Name=HDA Intel: CX20561 Analog (hw:0,0) MaxInChannels=2 MaxOutChannels=4 DefaultSampleRate=44100
Id=2 Name=USB Audio CODEC: USB Audio (hw:1,0) MaxInChannels=2 MaxOutChannels=2 DefaultSampleRate=44100
Id=3 Name=sysdefault MaxInChannels=128 MaxOutChannels=128 DefaultSampleRate=44100
Id=10 Name=pulse MaxInChannels=32 MaxOutChannels=32 DefaultSampleRate=44100
Id=12 Name=default MaxInChannels=32 MaxOutChannels=32 DefaultSampleRate=44100
$callback buffer count =5
callback buffer count =4
callback buffer count =3
callback buffer count =2
callback buffer count =1
callback buffer count =0
callback buffer count =9
callback buffer count =8
callback buffer count =7
0.0205011367798
callback buffer count =6
callback buffer count =5
callback buffer count =4
callback buffer count =3
callback buffer count =2
callback buffer count =1
callback buffer count =0
callback buffer count =9
0.0198979377747
callback buffer count =8
callback buffer count =7
callback buffer count =6
callback buffer count =5
callback buffer count =4
callback buffer count =3
callback buffer count =2
callback buffer count =1
callback buffer count =0
callback buffer count =9
0.0199301242828
callback buffer count =8
callback buffer count =7
callback buffer count =6
callback buffer count =5
callback buffer count =4.
.
callback buffer count =1
callback buffer count =0
callback buffer count =9
0.000388860702515
callback buffer count =8
callback buffer count =7
callback buffer count =6
callback buffer count =5
callback buffer count =4
callback buffer count =3
callback buffer count =2
callback buffer count =1
callback buffer count =0
callback buffer count =9
0.000384092330933
callback buffer count =8
callback buffer count =7
callback buffer count =6
.
.#! /usr/bin/python2.7
import pyaudio
import time
import sys
import numpy as np
from scipy.signal import butter, lfilter, freqz
# audio signal parameters
STREAM_CHANNELS = 2
STREAM_SAMPLE_RATE = 44100 #8000 #44100 #!!! sets time available for main program
STREAM_FORMAT = pyaudio.paFloat32
CHUNK_BUFFERS = 10 #!!! sets time available for main program
# bandpass filter parameters
BPLOWCUT = 900.0
BPHIGHCUT = 1100.0
BPORDER = 3
def GetFilterCoefficients():
global a, b
nyq = 0.5 * STREAM_SAMPLE_RATE
low = BPLOWCUT / nyq
high = BPHIGHCUT / nyq
b, a = butter(BPORDER, [low, high], btype='band')
return b, a
def BwBpFilter(data):
y = lfilter(b, a, data)
# y = lfilter(b, a, data, axis = (-1))
return y
def GetDefaultInputDevice():
CardId = p.get_default_input_device_info()["index"]
return CardId
def callback_2buffered (in_data, frame_count, time_info, status):
global RawData, DataReady, BufferCount, ChunkBuffer, FrameCount
# restart next data buffer
# data buffer = full
if BufferCount==0:
if DataReady==True:
print "!warning: 1 buffer may or may not have been lost!"
DataReady = True
# get first chunk
RawData = in_data
# save data = double buffered:
ChunkBuffer = RawData
BufferCount = CHUNK_BUFFERS
else:
# append one chunk
RawData = RawData + in_data
BufferCount = BufferCount - 1
FrameCount = frame_count
print "callback buffer count =%d"%BufferCount
return (in_data, pyaudio.paContinue)
# main program
print ""
print "full-duplex record/play through default soundcard"
print ""
p=pyaudio.PyAudio()
CardId=GetDefaultInputDevice()
print "Card id = %d"%CardId
#start with a full cycle of buffering & initialize RawData
BufferCount = 0
DataReady = False
InputStream = p.open(
format=STREAM_FORMAT,
channels=STREAM_CHANNELS,
rate = STREAM_SAMPLE_RATE,
input_device_index=CardId,
input=True,
output=False,
stream_callback=callback_2buffered)
InputStream.start_stream()
# wait for valid data to initialize arrays
GetFilterCoefficients()
time.sleep(2)
while InputStream.is_active():
if DataReady==True:
start = time.time()
NumArray = np.fromstring(ChunkBuffer, dtype=np.float32)
NumArrayRows = NumArray.size / STREAM_CHANNELS
#print "main loop, num array =%d"%NumArrayRows
NumArray = np.reshape(NumArray, (NumArrayRows, STREAM_CHANNELS))
NumArray = BwBpFilter(NumArray)
elapsed = time.time() - start
print elapsed
# send data done to callback
DataReady = False
#time.sleep(0.1)
InputStream.stop_stream()
InputStream.close()
p.terminate()![[Bild: 34_levelmeter.png]](https://stromrichter.org/d-amp/images/forumuploadpics/34_levelmeter.png)
Zitat:Original geschrieben von voltwide
Lieber Hoppi
bitte benutze diesen Parallelthread, damit der Original-Thread möglichst lesbar bleibt.
Wie man Python ausspricht?
Ganz einfach: Genauso wie Monty Python, nur ohne Monty!
An die hatte der Sprachschöpfer nämlich damals gedacht!
![[Bild: 38_plotint.png]](https://stromrichter.org/d-amp/images/forumuploadpics/38_plotint.png)
![[Bild: 76_plotext0.png]](https://stromrichter.org/d-amp/images/forumuploadpics/76_plotext0.png)
![[Bild: 84_plotext1.png]](https://stromrichter.org/d-amp/images/forumuploadpics/84_plotext1.png)
.
#! /usr/bin/python2.7
"""
Tongenerator mit PC, Soundkarte und Python
Erzeugt Dauertoene aus einer Tabelle von der Groesse eines chunks (typ 1024 samples).
Die Soundkarte uebernimmt zyklisch diese chunks und haengt sie aneinander.
Nur in seltenen Ausnahmefaellen passt ein ganzzahliges Vielfaches einer Schwingungsperiode
genau in den chunk-Puffer, so dass an den Stossstellen zweier chunks immer ein Phasensprung statt findet.
Derartige Phasenspruenge sind stoerend, sie werden gehoermaessig sofort wahr genommen.
Die dynamische Phasenkorrektur verschiebt den Startpunkt im chunkbuffer immer passend,
so dass nahtlose Uebergaenge zustande kommen.
Der Kode ist ausgelegt fuer geringstmoegliche CPU-Last.
Massgeblich dafuer ist die Tonausgabe per callback-Funktion,
die Phasenkorrektur durch dynamische Indizierung des chunk-Puffers
und das mittels sleep-Anweisung ausgeduennte polling.
Am Lenovo X200 wurde 1% Zusatzlast bei 44,1kHz Abstastrate gemessen.
Waehrend die Soundkarte auf dem linken Ausgang Toene ausgibt,
kann audicity diese simultan ueber den Eingang aufzeichen
(kabelverbindung Ausgang-Eingang links).
Auf diese Weise wurden Tonschnipsel zur Dokumentation erstellt.
Zur Laufzeit koennen Frequenz und Amplitude modifiziert werden
durch Ueberschreiben des chunk-Puffers.
Bevor es losgeht
Div Python-Pakete installieren
Default Soundkarte einstellen ueber die PulsAudio-GUI.
Oder PulseAudio killen und direkt ueber Alsa gehen.
Dazu muss im home-Verzeichnis eine passende Datei ".asoundrc" abgelegt werden.
Getestet mit Lenovo X200, Behringer UCA 202 unter ubuntu 14.04, ohne PulseAudio
Stand 2014-10-27
"""
import pyaudio
import time
import numpy as np
# Konfiguration
# Audiosignal-Parameter
STREAM_CHANNELS = 1
STREAM_SAMPLE_RATE = 44100
STREAM_FORMAT = pyaudio.paInt16
# Tongenerator
TONE_FQ = 1000.0
TONE_LEVEL = 0.5
CHUNK_LEN = 1024
# callback Funktion
def callback_tonegen (out_data, frame_count, time_info, status):
global NextChunk
if NextChunk == True:
print "!Fehler:output-chunk verloren!"
# Handshake-Uebergabe an Hauptprogramm
NextChunk = True
# PhasenSprungKorrektur durch ArrayIndexOffset ("slicing")
Data = DoubleChunkBuf[PhaseIdx::]
return (Data, pyaudio.paContinue)
# Befuelle Chunk-Puffer mit Sinus-Schwingungen
def DoubleChunkInit(fq, level, samplerate, chunklen):
maxphase = 2 * np.pi * fq * chunklen / samplerate
phaseofs = np.fmod(maxphase,(2 * np.pi))
x = np.arange (2*chunklen)
doublechunkbuf = np.array(x)
xcoeff = 2 * np.pi * fq / samplerate
doublechunkbuf = level*2**15*np.sin(x * xcoeff)
doublechunkbuf = doublechunkbuf.astype(np.int16)
return(doublechunkbuf,phaseofs,maxphase)
# Hauptprogramm Initialisierungen
Phase = 0.0
PhaseIdx = 0
NextChunk = False
DoubleChunkBuf, PhaseOfs, MaxPhase = DoubleChunkInit(TONE_FQ, TONE_LEVEL, STREAM_SAMPLE_RATE, CHUNK_LEN)
# initialisiere die Soundkarte
p = pyaudio.PyAudio()
# starte Audio-Stream
OutputStream = p.open(
format=STREAM_FORMAT,
channels=STREAM_CHANNELS,
rate = STREAM_SAMPLE_RATE,
input=False,
output=True,
stream_callback=callback_tonegen)
OutputStream.start_stream()
# Hauptprogrammschleife
while OutputStream.is_active():
if NextChunk == True:
NextChunk = False
# Versatz fuer PhasenSprungKorrektur
Phase += PhaseOfs
Phase = Phase % (2 * np.pi) #!!!
PhaseIdx = int(Phase * CHUNK_LEN / (MaxPhase))
time.sleep(0.001)
# Programmende
OutputStream.stop_stream()
OutputStream.close()
p.terminate()