Fix animations: renamed prefab from CustomSTT to TaSTT

Also:
* Check in toggle on/off animations
* Add toggle parameter
* libunity bug: getUniqueId() was calling allocateId() incorrectly
* Remove osc_ctrl `client` global
* Fix transcribe.py text encoding

1 files changed, 203 insertions, 187 deletions

diff --git a/transcribe.py b/transcribe.py
index d0e3574..dc36541 100644
--- a/transcribe.py
+++ b/transcribe.py
@@ -1,187 +1,203 @@
-#!/usr/bin/env python3

-

-import copy

-import fileinput

-import os

-import osc_ctrl

-# python3 -m pip install pyaudio

-import pyaudio

-import threading

-import time

-import wave

-# python3 -m pip install git+https://github.com/openai/whisper.git

-# python3 -m pip install torch -f https://download.pytorch.org/whl/torch_stable.html

-import whisper

-

-class AudioState:

-    CHUNK = 1024

-    FORMAT = pyaudio.paInt16

-    CHANNELS = 1

-    # This matches the framerate expected by whisper.

-    RATE = 16000

-

-    # The maximum length that recordAudio() will put into frames before it

-    # starts dropping from the start.

-    MAX_LENGTH_S = 90

-    # The minimum length that recordAudio() will wait for before saving audio.

-    MIN_LENGTH_S = 3

-

-    # PyAudio object

-    p = None

-

-    # PyAudio stream object

-    stream = None

-

-    frames = []

-    frames_lock = threading.Lock()

-

-    text = ""

-    text_lock = threading.Lock()

-

-    record_audio = True

-    transcribe_audio = True

-    send_audio = True

-

-def getMicStream():

-    audio_state = AudioState()

-    audio_state.p = pyaudio.PyAudio()

-

-    info = audio_state.p.get_host_api_info_by_index(0)

-    numdevices = info.get('deviceCount')

-

-    print("Finding index mic...")

-    got_match = False

-    device_index = -1

-    mic_str = "Focusrite"

-    index_str = "Digital Audio Interface"

-    target_str = mic_str

-    while got_match == False:

-        for i in range(0, numdevices):

-            if (audio_state.p.get_device_info_by_host_api_device_index(0, i).get('maxInputChannels')) > 0:

-                device_name = audio_state.p.get_device_info_by_host_api_device_index(0, i).get('name')

-                print("Input Device id ", i, " - ", device_name)

-                if target_str in device_name:

-                    print("Got match: {}".format(device_name))

-                    device_index = i

-                    got_match = True

-                    break

-        if got_match == False:

-            print("No match, sleeping")

-            time.sleep(3)

-

-    audio_state.stream = audio_state.p.open(format=audio_state.FORMAT,

-            channels=audio_state.CHANNELS, rate=audio_state.RATE,

-            input=True, frames_per_buffer=audio_state.CHUNK,

-            input_device_index=device_index)

-

-    return audio_state

-

-# Continuously records audio as long as audio_state.record_audio is set.

-def recordAudio(audio_state):

-    print("Recording audio")

-    while audio_state.record_audio:

-        data = audio_state.stream.read(audio_state.CHUNK)

-

-        audio_state.frames_lock.acquire()

-        audio_state.frames.append(data)

-        max_frames = int(audio_state.RATE * audio_state.MAX_LENGTH_S / audio_state.CHUNK)

-        if len(audio_state.frames) > max_frames:

-            audio_state.frames = audio_state.frames[-1 * max_frames :]

-        audio_state.frames_lock.release()

-

-    print("Done recording")

-

-# Saves audio. recordAudio() may continue running while this takes place.

-def saveAudio(audio_state, filename):

-    min_frames = int(audio_state.RATE * audio_state.MIN_LENGTH_S / audio_state.CHUNK)

-    if len(audio_state.frames) < min_frames:

-        return

-

-    wf = wave.open(filename, 'wb')

-    wf.setnchannels(audio_state.CHANNELS)

-    wf.setsampwidth(audio_state.p.get_sample_size(audio_state.FORMAT))

-    wf.setframerate(audio_state.RATE)

-

-    audio_state.frames_lock.acquire()

-    frames = copy.deepcopy(audio_state.frames)

-    audio_state.frames_lock.release()

-

-    wf.writeframes(b''.join(frames))

-    wf.close()

-

-def resetAudio(audio_state):

-    audio_state.frames_lock.acquire()

-    audio_state.frames = []

-    audio_state.frames_lock.release()

-

-# Transcribe the audio recorded in a file.

-def transcribe(model, filename):

-    result = whisper.transcribe(model=model, audio=filename, language="en")

-    return result["text"]

-

-def transcribeAudio(audio_state, model):

-    while audio_state.transcribe_audio == True:

-        saveAudio(audio_state, "audio.wav")

-

-        if not os.path.isfile("audio.wav"):

-            time.sleep(0.1)

-            continue

-

-        print("Beginning transcription")

-        text = transcribe(model, "audio.wav")

-

-        audio_state.text_lock.acquire()

-        audio_state.text = text

-        audio_state.text_lock.release()

-

-        print("Transcription: {}".format(audio_state.text))

-

-        # Pace this out

-        time.sleep(0.2)

-

-def sendAudio(audio_state):

-    tx_state = osc_ctrl.OscTxState()

-    while audio_state.send_audio == True:

-        audio_state.text_lock.acquire()

-        text = copy.deepcopy(audio_state.text)

-        audio_state.text_lock.release()

-

-        osc_ctrl.sendMessageLazy(text, tx_state)

-

-        # Pace this out

-        time.sleep(0.05)

-

-if __name__ == "__main__":

-    if os.path.isfile("audio.wav"):

-        os.remove("audio.wav")

-

-    audio_state = getMicStream()

-

-    record_audio_thd = threading.Thread(target = recordAudio, args = [audio_state])

-    record_audio_thd.daemon = True

-    record_audio_thd.start()

-

-    print("Safe to start talking")

-

-    model = whisper.load_model("base")

-

-    transcribe_audio_thd = threading.Thread(target = transcribeAudio, args = [audio_state, model])

-    transcribe_audio_thd.daemon = True

-    transcribe_audio_thd.start()

-

-    #send_audio_thd = threading.Thread(target = sendAudio, args = [audio_state])

-    #send_audio_thd.daemon = True

-    #send_audio_thd.start()

-

-    print("Press enter to start a new message")

-    for line in fileinput.input():

-        resetAudio(audio_state)

-        if "exit" in line or "quit" in line:

-            break

-

-    print("Joining threads")

-    audio_state.record_audio = False

-    audio_state.transcribe_audio = False

-    record_audio_thd.join()

-    transcribe_audio_thd.join()

-

+#!/usr/bin/env python3
+
+import argparse
+import copy
+import os
+import osc_ctrl
+# python3 -m pip install pyaudio
+import pyaudio
+import sys
+import threading
+import time
+import wave
+# python3 -m pip install git+https://github.com/openai/whisper.git
+# python3 -m pip install torch -f https://download.pytorch.org/whl/torch_stable.html
+from whisper import transcribe as whisper_transcribe
+from whisper import load_model as whisper_load_model
+
+class AudioState:
+    CHUNK = 1024
+    FORMAT = pyaudio.paInt16
+    CHANNELS = 1
+    # This matches the framerate expected by whisper.
+    RATE = 16000
+
+    # The maximum length that recordAudio() will put into frames before it
+    # starts dropping from the start.
+    MAX_LENGTH_S = 90
+    # The minimum length that recordAudio() will wait for before saving audio.
+    MIN_LENGTH_S = 3
+
+    # PyAudio object
+    p = None
+
+    # PyAudio stream object
+    stream = None
+
+    frames = []
+    frames_lock = threading.Lock()
+
+    text = ""
+    text_lock = threading.Lock()
+
+    record_audio = True
+    transcribe_audio = True
+    send_audio = True
+
+    osc_client = osc_ctrl.getClient()
+
+def getMicStream(which_mic):
+    audio_state = AudioState()
+    audio_state.p = pyaudio.PyAudio()
+
+    print("Finding index mic...")
+    got_match = False
+    device_index = -1
+    focusrite_str = "Focusrite"
+    index_str = "Digital Audio Interface"
+    if which_mic == "index":
+        target_str = index_str
+    elif which_mic == "focusrite":
+        target_str = focusrite_str
+    else:
+        raise Exception("Unrecognized mic requested: {}".format(which_mic))
+    while got_match == False:
+        info = audio_state.p.get_host_api_info_by_index(0)
+        numdevices = info.get('deviceCount')
+
+        for i in range(0, numdevices):
+            if (audio_state.p.get_device_info_by_host_api_device_index(0, i).get('maxInputChannels')) > 0:
+                device_name = audio_state.p.get_device_info_by_host_api_device_index(0, i).get('name')
+                print("Input Device id ", i, " - ", device_name)
+                if target_str in device_name:
+                    print("Got match: {}".format(device_name))
+                    device_index = i
+                    got_match = True
+                    break
+        if got_match == False:
+            print("No match, sleeping")
+            time.sleep(3)
+
+    audio_state.stream = audio_state.p.open(format=audio_state.FORMAT,
+            channels=audio_state.CHANNELS, rate=audio_state.RATE,
+            input=True, frames_per_buffer=audio_state.CHUNK,
+            input_device_index=device_index)
+
+    return audio_state
+
+# Continuously records audio as long as audio_state.record_audio is set.
+def recordAudio(audio_state):
+    print("Recording audio")
+    while audio_state.record_audio:
+        data = audio_state.stream.read(audio_state.CHUNK)
+
+        audio_state.frames_lock.acquire()
+        audio_state.frames.append(data)
+        max_frames = int(audio_state.RATE * audio_state.MAX_LENGTH_S / audio_state.CHUNK)
+        if len(audio_state.frames) > max_frames:
+            audio_state.frames = audio_state.frames[-1 * max_frames :]
+        audio_state.frames_lock.release()
+
+    print("Done recording")
+
+# Saves audio. recordAudio() may continue running while this takes place.
+def saveAudio(audio_state, filename):
+    min_frames = int(audio_state.RATE * audio_state.MIN_LENGTH_S / audio_state.CHUNK)
+    if len(audio_state.frames) < min_frames:
+        return
+
+    wf = wave.open(filename, 'wb')
+    wf.setnchannels(audio_state.CHANNELS)
+    wf.setsampwidth(audio_state.p.get_sample_size(audio_state.FORMAT))
+    wf.setframerate(audio_state.RATE)
+
+    audio_state.frames_lock.acquire()
+    frames = copy.deepcopy(audio_state.frames)
+    audio_state.frames_lock.release()
+
+    wf.writeframes(b''.join(frames))
+    wf.close()
+
+def resetAudio(audio_state):
+    audio_state.frames_lock.acquire()
+    audio_state.frames = []
+    audio_state.frames_lock.release()
+
+# Transcribe the audio recorded in a file.
+def transcribe(model, filename):
+    result = whisper_transcribe(model=model, audio=filename, language="en")
+    return result["text"]
+
+def transcribeAudio(audio_state, model):
+    while audio_state.transcribe_audio == True:
+        saveAudio(audio_state, "audio.wav")
+
+        if not os.path.isfile("audio.wav"):
+            time.sleep(0.1)
+            continue
+
+        print("Beginning transcription")
+        text = transcribe(model, "audio.wav")
+
+        audio_state.text_lock.acquire()
+        audio_state.text = text
+        audio_state.text_lock.release()
+
+        print("Transcription: {}".format(audio_state.text))
+
+        # Pace this out
+        time.sleep(0.2)
+
+def sendAudio(audio_state):
+    tx_state = osc_ctrl.OscTxState()
+    while audio_state.send_audio == True:
+        audio_state.text_lock.acquire()
+        text = copy.deepcopy(audio_state.text)
+        audio_state.text_lock.release()
+
+        osc_ctrl.sendMessageLazy(audio_state.osc_client, text, tx_state)
+
+        # Pace this out
+        time.sleep(0.05)
+
+if __name__ == "__main__":
+    parser = argparse.ArgumentParser()
+    parser.add_argument("--mic", type=str, help="Which mic to use. Options: index, focusrite. Default: index")
+    args = parser.parse_args()
+
+    if not args.mic:
+        args.mic = "index"
+
+    if os.path.isfile("audio.wav"):
+        os.remove("audio.wav")
+
+    audio_state = getMicStream(args.mic)
+
+    record_audio_thd = threading.Thread(target = recordAudio, args = [audio_state])
+    record_audio_thd.daemon = True
+    record_audio_thd.start()
+
+    print("Safe to start talking")
+
+    model = whisper_load_model("base")
+
+    transcribe_audio_thd = threading.Thread(target = transcribeAudio, args = [audio_state, model])
+    transcribe_audio_thd.daemon = True
+    transcribe_audio_thd.start()
+
+    send_audio_thd = threading.Thread(target = sendAudio, args = [audio_state])
+    send_audio_thd.daemon = True
+    send_audio_thd.start()
+
+    print("Press enter to start a new message")
+    for line in sys.stdin:
+        resetAudio(audio_state)
+        if "exit" in line or "quit" in line:
+            break
+
+    print("Joining threads")
+    audio_state.record_audio = False
+    audio_state.transcribe_audio = False
+    record_audio_thd.join()
+    transcribe_audio_thd.join()
+