moving things around and adding examples for mic hat code

mic_hat_examples/recording_examples/get_device_index.py

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+import pyaudio
+
+p = pyaudio.PyAudio()
+info = p.get_host_api_info_by_index(0)
+numdevices = info.get('deviceCount')
+
+for i in range(0, numdevices):
+    if (p.get_device_info_by_host_api_device_index(0, i).get('maxInputChannels')) > 0:
+        print("Input Device id ", i, " - ", p.get_device_info_by_host_api_device_index(0, i).get('name'))
+    else:
+        print("Output Device id ", i, " - ", p.get_device_info_by_host_api_device_index(0, i).get('name'))

mic_hat_examples/recording_examples/play.py

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+import pyaudio
+import wave
+import sys
+
+# length of data to read.
+chunk = 1024
+RESPEAKER_INDEX = 1
+# validation. If a wave file hasn't been specified, exit.
+if len(sys.argv) < 2:
+    print("Plays a wave file.\n\n" + "Usage: %s filename.wav" % sys.argv[0])
+    sys.exit(-1)
+
+# open the file for reading.
+wf = wave.open(sys.argv[1], 'rb')
+
+# create an audio object
+p = pyaudio.PyAudio()
+
+# open stream based on the wave object which has been input.
+stream = p.open(format = p.get_format_from_width(wf.getsampwidth()),
+                channels = wf.getnchannels(),
+                rate = wf.getframerate(),
+                output = True,
+                output_device_index = RESPEAKER_INDEX)
+
+# read data (based on the chunk size)
+data = wf.readframes(chunk)
+# play stream (looping from beginning of file to the end)
+while data:
+    # writing to the stream is what *actually* plays the sound.
+    stream.write(data)
+    data = wf.readframes(chunk)
+
+# cleanup stuff.
+stream.close()    
+p.terminate()

mic_hat_examples/recording_examples/record.py

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+import pyaudio
+import wave
+
+RESPEAKER_RATE = 16000
+RESPEAKER_CHANNELS = 2 
+RESPEAKER_WIDTH = 2
+# run getDeviceInfo.py to get index
+RESPEAKER_INDEX = 1  # refer to input device id
+CHUNK = 1024
+RECORD_SECONDS = 5
+WAVE_OUTPUT_FILENAME = "output.wav"
+
+p = pyaudio.PyAudio()
+
+stream = p.open(
+            rate=RESPEAKER_RATE,
+            format=p.get_format_from_width(RESPEAKER_WIDTH),
+            channels=RESPEAKER_CHANNELS,
+            input=True,
+            input_device_index=RESPEAKER_INDEX,)
+
+print("* recording")
+
+frames = []
+
+for i in range(0, int(RESPEAKER_RATE / CHUNK * RECORD_SECONDS)):
+    data = stream.read(CHUNK)
+    frames.append(data)
+
+print("* done recording")
+
+stream.stop_stream()
+stream.close()
+p.terminate()
+
+wf = wave.open(WAVE_OUTPUT_FILENAME, 'wb')
+wf.setnchannels(RESPEAKER_CHANNELS)
+wf.setsampwidth(p.get_sample_size(p.get_format_from_width(RESPEAKER_WIDTH)))
+wf.setframerate(RESPEAKER_RATE)
+wf.writeframes(b''.join(frames))
+wf.close()

mic_hat_examples/recording_examples/record_one_channel.py

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+import pyaudio
+import wave
+import numpy as np
+
+RESPEAKER_RATE = 16000
+RESPEAKER_CHANNELS = 2
+RESPEAKER_WIDTH = 2
+# run getDeviceInfo.py to get index
+RESPEAKER_INDEX = 1  # refer to input device id
+CHUNK = 1024
+RECORD_SECONDS = 3
+WAVE_OUTPUT_FILENAME = "output_one_channel.wav"
+
+p = pyaudio.PyAudio()
+
+stream = p.open(
+            rate=RESPEAKER_RATE,
+            format=p.get_format_from_width(RESPEAKER_WIDTH),
+            channels=RESPEAKER_CHANNELS,
+            input=True,
+            input_device_index=RESPEAKER_INDEX,)
+
+print("* recording")
+
+frames = [] 
+
+for i in range(0, int(RESPEAKER_RATE / CHUNK * RECORD_SECONDS)):
+    data = stream.read(CHUNK)
+    # extract channel 0 data from 2 channels, if you want to extract channel 1, please change to [1::2]
+    a = np.fromstring(data,dtype=np.int16)[0::2]
+    frames.append(a.tostring())
+
+print("* done recording")
+
+stream.stop_stream()
+stream.close()
+p.terminate()
+
+wf = wave.open(WAVE_OUTPUT_FILENAME, 'wb')
+wf.setnchannels(1)
+wf.setsampwidth(p.get_sample_size(p.get_format_from_width(RESPEAKER_WIDTH)))
+wf.setframerate(RESPEAKER_RATE)
+wf.writeframes(b''.join(frames))
+wf.close()
+