move to sounddevice
This commit is contained in:
Ajay Raj
parent
505bba2e07
commit
565bd5b3a1
4 changed files with 44 additions and 45 deletions
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@ -5,7 +5,7 @@ import signal
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from vocode.conversation import Conversation
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from vocode.helpers import create_microphone_input_and_speaker_output
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from vocode.models.transcriber import DeepgramTranscriberConfig
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from vocode.models.agent import ChatGPTAgentConfig, RESTfulUserImplementedAgentConfig, WebSocketUserImplementedAgentConfig
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from vocode.models.agent import ChatGPTAgentConfig, RESTfulUserImplementedAgentConfig, WebSocketUserImplementedAgentConfig, EchoAgentConfig
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from vocode.models.synthesizer import AzureSynthesizerConfig
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from vocode.user_implemented_agent.restful_agent import RESTfulAgent
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@ -14,19 +14,15 @@ logging.root.setLevel(logging.INFO)
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if __name__ == "__main__":
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microphone_input, speaker_output = create_microphone_input_and_speaker_output(use_first_available_device=False)
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microphone_input, speaker_output = create_microphone_input_and_speaker_output(use_default_devices=True)
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conversation = Conversation(
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input_device=microphone_input,
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output_device=speaker_output,
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transcriber_config=DeepgramTranscriberConfig.from_input_device(microphone_input),
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agent_config=RESTfulUserImplementedAgentConfig(
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agent_config=EchoAgentConfig(
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initial_message="Hello!",
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generate_responses=False,
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respond=RESTfulUserImplementedAgentConfig.EndpointConfig(
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url="http://a6eb64f4a9b7.ngrok.io/respond",
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method="POST"
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)
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),
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synthesizer_config=AzureSynthesizerConfig.from_output_device(speaker_output)
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)
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@ -1,4 +1,4 @@
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import pyaudio
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import sounddevice as sd
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from .input_device.microphone_input import MicrophoneInput
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from .output_device.speaker_output import SpeakerOutput
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import logging
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@ -11,20 +11,18 @@ def _get_device_prompt(device_infos: list[dict]) -> str:
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Choice: """.format(
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"\n".join(f"{index}: {device['name']}" for index, device in enumerate(device_infos)))
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def create_microphone_input_and_speaker_output(use_first_available_device=False, mic_sampling_rate=None, speaker_sampling_rate=None) -> tuple[MicrophoneInput, SpeakerOutput]:
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pa = pyaudio.PyAudio()
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num_devices = pa.get_device_count()
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devices = list(map(pa.get_device_info_by_index, range(num_devices)))
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input_device_infos = list(filter(lambda device: device['maxInputChannels'] > 0, devices))
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output_device_infos = list(filter(lambda device: device['maxOutputChannels'] > 0, devices))
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if use_first_available_device:
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input_device_info = input_device_infos[0]
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output_device_info = output_device_infos[0]
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def create_microphone_input_and_speaker_output(use_default_devices=False, mic_sampling_rate=None, speaker_sampling_rate=None) -> tuple[MicrophoneInput, SpeakerOutput]:
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device_infos = sd.query_devices()
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input_device_infos = list(filter(lambda device_info: device_info['max_input_channels'] > 0, device_infos))
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output_device_infos = list(filter(lambda device_info: device_info['max_output_channels'] > 0, device_infos))
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if use_default_devices:
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input_device_info = sd.query_devices(kind='input')
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output_device_info = sd.query_devices(kind='output')
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else:
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input_device_info = input_device_infos[int(input(_get_device_prompt(input_device_infos)))]
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output_device_info = output_device_infos[int(input(_get_device_prompt(output_device_infos)))]
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logger.info("Using microphone input device: %s", input_device_info['name'])
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microphone_input = MicrophoneInput(pa, input_device_info, sampling_rate=mic_sampling_rate)
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microphone_input = MicrophoneInput(input_device_info, sampling_rate=mic_sampling_rate)
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logger.info("Using speaker output device: %s", output_device_info['name'])
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speaker_output = SpeakerOutput(pa, output_device_info, sampling_rate=speaker_sampling_rate)
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speaker_output = SpeakerOutput(output_device_info, sampling_rate=speaker_sampling_rate)
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return microphone_input, speaker_output
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@ -1,6 +1,8 @@
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import pyaudio
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import sounddevice as sd
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import numpy as np
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from typing import Optional
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import queue
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import wave
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from .base_input_device import BaseInputDevice
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from ..models.audio_encoding import AudioEncoding
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@ -10,25 +12,29 @@ class MicrophoneInput(BaseInputDevice):
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DEFAULT_SAMPLING_RATE = 44100
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DEFAULT_CHUNK_SIZE = 2048
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def __init__(self, pa: pyaudio.PyAudio, device_info: dict, sampling_rate: int = None, chunk_size: int = DEFAULT_CHUNK_SIZE):
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def __init__(self, device_info: dict, sampling_rate: int = None, chunk_size: int = DEFAULT_CHUNK_SIZE, microphone_gain: int = 1):
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self.device_info = device_info
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sampling_rate = sampling_rate or (self.device_info.get('defaultSampleRate', self.DEFAULT_SAMPLING_RATE))
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sampling_rate = sampling_rate or (self.device_info.get('default_samplerate', self.DEFAULT_SAMPLING_RATE))
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super().__init__(int(sampling_rate), AudioEncoding.LINEAR16, chunk_size)
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self.pa = pa
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self.stream = pa.open(
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format=pyaudio.paInt16,
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self.stream = sd.InputStream(
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dtype=np.int16,
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channels=1,
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rate=self.sampling_rate,
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input=True,
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frames_per_buffer=self.chunk_size,
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input_device_index=int(self.device_info['index']),
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stream_callback=self._stream_callback
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samplerate=self.sampling_rate,
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blocksize=self.chunk_size,
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device=int(self.device_info['index']),
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callback=self._stream_callback
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)
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self.stream.start()
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self.queue = queue.Queue()
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self.microphone_gain = microphone_gain
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def _stream_callback(self, in_data, *_args):
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self.queue.put_nowait(in_data)
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return (None, pyaudio.paContinue)
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def _stream_callback(self, in_data: np.ndarray[np.int16], *_args):
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if self.microphone_gain > 1:
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in_data = in_data * (2 ^ self.microphone_gain)
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else:
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in_data = in_data // (2 ^ self.microphone_gain)
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audio_bytes = in_data.tobytes()
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self.queue.put_nowait(audio_bytes)
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def get_audio(self) -> Optional[bytes]:
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try:
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@ -1,4 +1,5 @@
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import pyaudio
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import sounddevice as sd
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import numpy as np
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from .base_output_device import BaseOutputDevice
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from ..models.audio_encoding import AudioEncoding
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@ -7,22 +8,20 @@ class SpeakerOutput(BaseOutputDevice):
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DEFAULT_SAMPLING_RATE = 44100
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def __init__(self, pa: pyaudio.PyAudio, device_info: dict, sampling_rate: int = None, audio_encoding: AudioEncoding = AudioEncoding.LINEAR16):
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def __init__(self, device_info: dict, sampling_rate: int = None, audio_encoding: AudioEncoding = AudioEncoding.LINEAR16):
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self.device_info = device_info
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sampling_rate = sampling_rate or int(self.device_info.get('defaultSampleRate', self.DEFAULT_SAMPLING_RATE))
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sampling_rate = sampling_rate or int(self.device_info.get('default_samplerate', self.DEFAULT_SAMPLING_RATE))
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super().__init__(sampling_rate, audio_encoding)
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self.pa = pa
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self.stream = self.pa.open(
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output=True,
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self.stream = sd.OutputStream(
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channels=1,
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rate=self.sampling_rate,
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format=pyaudio.paInt16,
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output_device_index=int(self.device_info['index'])
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samplerate=self.sampling_rate,
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dtype=np.int16,
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device=int(self.device_info['index'])
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)
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self.stream.start()
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async def send_async(self, chunk):
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self.stream.write(chunk)
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self.stream.write(np.frombuffer(chunk, dtype=np.int16))
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def terminate(self):
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self.stream.close()
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self.pa.close()
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self.stream.close()
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