Update with new postprocessor

Author: tippesi

README.md

@@ -1 +1,9 @@
 # Python Audio
+## What you need
+- Python 3.6.x installed (PyAudio won't work with 3.7.x)
+- PyAudio installed
+## What you get
+- Tiny framework which allows to play with audio effects
+- Easy to integratge postprocessing steps per audio stream
+## Example
+Have a look at main.py

audio.py

@@ -12,6 +12,7 @@ class Audio():
     """
 
     """
+
     def __init__(self, filename, chunksize, paudio):
 
         filestream = wave.open(filename, "rb")
@@ -23,6 +24,7 @@ def __init__(self, filename, chunksize, paudio):
 
         self.progress = 0.0
         self.pitch = 1.0
+        self.loop = False
         self.channels = filestream.getnchannels()
         self.frequency = filestream.getframerate()
         self.postprocessing = []
@@ -62,7 +64,14 @@ def __del__(self):
         self._audiostream.close()
 
     def update(self):
-        self.progress = self.progress % float(len(self._audiodata))        
+        # Check if playback should be in a loop
+        if self.loop:
+            self.progress = self.progress % float(len(self._audiodata))
+        else:
+            if self.progress >= len(self._audiodata):
+                return
+
+        # Depending on the pitch we want to process a different number of samples
         readout = self.pitch * float(self._chunksize)
         index = math.floor(self.progress)
         count = math.ceil(readout)
@@ -75,12 +84,11 @@ def update(self):
         for postprocessor in self.postprocessing:
             subdata = postprocessor.apply(subdata, self.channels, self.frequency)
 
-        # Check values are in their specified range
+        # Check if values are in their specified range
         for i in range(0, len(subdata)):
             subdata[i] = max(min(subdata[i], self._maxvalue), -self._maxvalue)
 
         fmt = self._endianness + self._type * len(subdata)
         audiodata = struct.pack(fmt, *subdata)
-        print(audiodata)
         self._audiostream.write(audiodata)
         self.progress += readout

main.py

@@ -1,14 +1,14 @@
-import audio
+from audio import Audio
 from postprocessing.volume_postprocessing import VolumePostprocessing
 from postprocessing.channel_sine_postprocessing import ChannelSinePostprocessing
 import pyaudio
 
 paudio = pyaudio.PyAudio()
 
-audio = audio.Audio("MenuTheme2_final.wav", 1024, paudio)
+audio = Audio("MenuTheme2_final.wav", 2048, paudio)
 
 # audio.postprocessing.append(VolumePostprocessing(1.0))
-audio.postprocessing.append(ChannelSinePostprocessing(0.1))
+audio.postprocessing.append(ChannelSinePostprocessing(0.5))
 
 while True:
     audio.update()

postprocessing/channel_sine_postprocessing.py

@@ -20,15 +20,10 @@ def apply(self, data, channels, frequency):
         leftVolume = 0.5 * math.sin(deltatime) + 0.5
         rightVolume = 1.0 - leftVolume
 
-        # print(leftVolume)
-
         for i in range(0, len(data)):
-            """
             if i % 2:
                 data[i] = int(float(data[i]) * leftVolume)
             else:
                 data[i] = int(float(data[i]) * rightVolume)
-            """
-            data[i] = data[i] & 255
 
         return data