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https://github.com/maglore9900/max_headroom.git
synced 2025-06-06 19:45:31 +00:00
pitch / stutter tweaking
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maglore9900
parent
28872342ee
commit
46e61fc9fe
4
main.py
4
main.py
@ -11,10 +11,10 @@ graph = agent.Agent()
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while True:
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text = sp.listen2()
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if text and "max" in text.lower():
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if text and "max" in text.lower() or text and "mac" in text.lower():
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if "exit" in text.lower():
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break
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response = loop.run_until_complete(graph.invoke_agent(text))
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if response:
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sp.glitch_stream_output2(response)
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sp.glitch_stream_output(response)
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@ -132,6 +132,8 @@ class Agent:
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self.sp.play()
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elif command == "pause":
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self.sp.pause()
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elif command == "stop":
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self.sp.pause()
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elif command == "next":
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self.sp.next_track()
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elif command == "previous":
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158
modules/speak.py
158
modules/speak.py
@ -30,43 +30,8 @@ class Speak:
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self.model = Model(self.model_path)
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self.recognizer = KaldiRecognizer(self.model, 16000)
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def max_headroom(self, text):
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data = {
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"text_input": str(text),
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"text_filtering": "standard",
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"character_voice_gen": "maxheadroom_00000045.wav",
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"narrator_enabled": "false",
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"narrator_voice_gen": "male_01.wav",
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"text_not_inside": "character",
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"language": "en",
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"output_file_name": "stream_output",
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"output_file_timestamp": "true",
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"autoplay": "false",
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"autoplay_volume": "0.8"
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}
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# Send the POST request to generate TTS
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response = requests.post(self.url, data=data)
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# Check if the request was successful
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if response.status_code == 200:
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# Parse the JSON response to get the file URL
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result = response.json()
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audio_url = result['output_file_url']
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# Download the audio file
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audio_response = requests.get(audio_url)
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output_path = os.path.abspath("tmp/output.wav")
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# Save the audio file locally
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with open(output_path, "wb") as f:
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f.write(audio_response.content)
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winsound.PlaySound(output_path, winsound.SND_FILENAME)
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else:
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print(f"Failed with status code {response.status_code}: {response.text}")
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self.engine.say(text)
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self.engine.runAndWait()
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#! listen with google
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def listen(self):
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with self.microphone as source:
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# Adjust for ambient noise
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@ -89,39 +54,15 @@ class Speak:
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print("Timeout. No speech detected.")
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return None
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# def listen2(self):
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# p = pyaudio.PyAudio()
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# stream = p.open(format=pyaudio.paInt16, channels=1, rate=16000, input=True, frames_per_buffer=8000)
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# stream.start_stream()
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# print("Listening...")
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# try:
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# while True:
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# data = stream.read(8000, exception_on_overflow=False)
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# filtered_data = nr.reduce_noise(y=frombuffer(data, dtype=int16), sr=16000).astype(int16).tobytes()
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# if self.recognizer.AcceptWaveform(filtered_data):
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# result = json.loads(self.recognizer.Result())
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# if result["text"]:
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# print(f"Recognized: {result['text']}")
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# return result['text']
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# else:
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# pass
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# except KeyboardInterrupt:
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# print("Stopping...")
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# finally:
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# stream.stop_stream()
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# stream.close()
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# p.terminate()
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#! listen with vosk
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def listen2(self, noise_threshold=500):
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p = pyaudio.PyAudio()
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stream = p.open(format=pyaudio.paInt16, channels=1, rate=16000, input=True, frames_per_buffer=8000)
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stream.start_stream()
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print("Listening...")
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count = 0
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try:
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while True:
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while count < 10:
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data = stream.read(8000, exception_on_overflow=False)
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filtered_data = nr.reduce_noise(y=frombuffer(data, dtype=int16), sr=16000).astype(int16).tobytes()
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@ -136,6 +77,7 @@ class Speak:
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return result['text']
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else:
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print(f"Ambient noise detected: RMS {rms_value} exceeds threshold {noise_threshold}")
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count += 1
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except KeyboardInterrupt:
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print("Stopping...")
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finally:
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@ -143,7 +85,6 @@ class Speak:
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stream.close()
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p.terminate()
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def stream_output(self, text):
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import urllib.parse
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# Example parameters
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@ -177,85 +118,10 @@ class Speak:
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if state in [vlc.State.Ended, vlc.State.Stopped, vlc.State.Error]:
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break
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time.sleep(1)
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def glitch_stream_output(self, text):
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def change_pitch(sound, octaves):
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val = random.randint(0, 10)
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if val == 1:
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new_sample_rate = int(sound.frame_rate * (2.0 ** octaves))
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return sound._spawn(sound.raw_data, overrides={'frame_rate': new_sample_rate}).set_frame_rate(sound.frame_rate)
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else:
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return sound
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# Example parameters
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voice = "maxheadroom_00000045.wav"
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language = "en"
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output_file = "stream_output.wav"
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# Encode the text for URL
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encoded_text = urllib.parse.quote(text)
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# Create the streaming URL
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streaming_url = f"http://localhost:7851/api/tts-generate-streaming?text={encoded_text}&voice={voice}&language={language}&output_file={output_file}"
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# Stream the audio data
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response = requests.get(streaming_url, stream=True)
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# Initialize PyAudio
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p = pyaudio.PyAudio()
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stream = None
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# Process the audio stream in chunks
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chunk_size = 1024 * 6 # Adjust chunk size if needed
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audio_buffer = b''
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for chunk in response.iter_content(chunk_size=chunk_size):
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audio_buffer += chunk
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if len(audio_buffer) < chunk_size:
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continue
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audio_segment = AudioSegment(
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data=audio_buffer,
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sample_width=2, # 2 bytes for 16-bit audio
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# frame_rate=44100, # Assumed frame rate, adjust as necessary
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frame_rate=24000, # Assumed frame rate, adjust as necessary
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channels=1 # Assuming mono audio
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)
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# Randomly adjust pitch
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# octaves = random.uniform(-0.5, 0.5)
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octaves = random.uniform(-0.5, 1)
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modified_chunk = change_pitch(audio_segment, octaves)
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if stream is None:
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# Define stream parameters
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stream = p.open(format=pyaudio.paInt16,
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channels=1,
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rate=modified_chunk.frame_rate,
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output=True)
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if random.random() < 0.001: # 1% chance to trigger stutter
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repeat_times = random.randint(2, 5) # Repeat 2 to 5 times
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for _ in range(repeat_times):
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stream.write(modified_chunk.raw_data)
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# Play the modified chunk
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stream.write(modified_chunk.raw_data)
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# Reset buffer
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audio_buffer = b''
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# Final cleanup
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if stream:
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stream.stop_stream()
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stream.close()
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p.terminate()
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def glitch_stream_output2(self, text):
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def change_pitch(sound, octaves):
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val = random.randint(0, 10)
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val = random.randint(0, 7)
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if val == 1:
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new_sample_rate = int(sound.frame_rate * (2.0 ** octaves))
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return sound._spawn(sound.raw_data, overrides={'frame_rate': new_sample_rate}).set_frame_rate(sound.frame_rate)
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@ -288,7 +154,7 @@ class Speak:
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stream = None
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# Process the audio stream in chunks
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chunk_size = 1024 * 6 # Adjust chunk size if needed
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chunk_size = 1024 * 7 # Adjust chunk size if needed
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audio_buffer = b''
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for chunk in response.iter_content(chunk_size=chunk_size):
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@ -305,7 +171,7 @@ class Speak:
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)
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# Randomly adjust pitch
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octaves = random.uniform(-1, 1)
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octaves = random.uniform(-0.5, 1.5)
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modified_chunk = change_pitch(audio_segment, octaves)
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if random.random() < 0.01: # 1% chance to trigger stutter
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@ -336,4 +202,8 @@ class Speak:
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p.terminate()
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except:
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self.engine.say(text)
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self.engine.runAndWait()
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self.engine.runAndWait()
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# sp = Speak()
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# sp.glitch_stream_output2("this is a test of pitch and stutter. test 1 2 3. I just need a long enough sentence to see the frequecy of sound changes.")
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@ -33,13 +33,6 @@ class Spotify:
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self.sp.pause_playback(device_id=device_id)
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except Exception as e:
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print(f"Failed to pause playback: {e}")
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def stop(self):
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try:
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device_id = self.get_active_device()
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self.sp.pause_playback(device_id=device_id)
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except Exception as e:
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print(f"Failed to stop playback: {e}")
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def next_track(self):
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try:
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