remove pyq goodbye model and rime synthesizer and fix environment loading

This commit is contained in:
Ajay Raj 2023-03-28 10:20:36 -07:00
commit 1dc7bc74c3
28 changed files with 143 additions and 285 deletions

View file

@ -1,10 +1,9 @@
import asyncio
import json
import logging
import os
from dotenv import load_dotenv
import websockets
from urllib.parse import urlencode
from vocode import getenv
from vocode.streaming.models.transcriber import AssemblyAITranscriberConfig
from vocode.streaming.models.websocket import AudioMessage
@ -14,9 +13,7 @@ from vocode.streaming.transcriber.base_transcriber import (
)
from vocode.streaming.models.audio_encoding import AudioEncoding
load_dotenv()
ASSEMBLY_AI_API_KEY = os.environ.get("ASSEMBLY_AI_API_KEY")
ASSEMBLY_AI_URL = "wss://api.assemblyai.com/v2/realtime/ws"
@ -27,6 +24,7 @@ class AssemblyAITranscriber(BaseTranscriber):
logger: logging.Logger = None,
):
super().__init__(transcriber_config)
self.api_key = getenv("ASSEMBLY_AI_API_KEY")
self._ended = False
self.is_ready = False
self.logger = logger or logging.getLogger(__name__)
@ -61,7 +59,7 @@ class AssemblyAITranscriber(BaseTranscriber):
async with websockets.connect(
URL,
extra_headers=(("Authorization", ASSEMBLY_AI_API_KEY),),
extra_headers=(("Authorization", self.api_key),),
ping_interval=5,
ping_timeout=20,
) as ws:

View file

@ -1,11 +1,8 @@
from dotenv import load_dotenv
from typing import Callable, Optional, Awaitable
from vocode.streaming.utils import convert_wav
from vocode.streaming.models.transcriber import EndpointingConfig, TranscriberConfig
load_dotenv()
class Transcription:
def __init__(

View file

@ -1,12 +1,11 @@
import asyncio
import json
import logging
import os
from dotenv import load_dotenv
import websockets
from websockets.client import WebSocketClientProtocol
import audioop
from urllib.parse import urlencode
from vocode import getenv
from vocode.streaming.transcriber.base_transcriber import (
BaseTranscriber,
@ -19,9 +18,7 @@ from vocode.streaming.models.transcriber import (
)
from vocode.streaming.models.audio_encoding import AudioEncoding
load_dotenv()
DEEPGRAM_API_KEY = os.environ.get("DEEPGRAM_API_KEY")
PUNCTUATION_TERMINATORS = [".", "!", "?"]
NUM_RESTARTS = 5
@ -33,6 +30,7 @@ class DeepgramTranscriber(BaseTranscriber):
logger: logging.Logger = None,
):
super().__init__(transcriber_config)
self.api_key = getenv("DEEPGRAM_API_KEY")
self.transcriber_config = transcriber_config
self._ended = False
self.warmed_up = False
@ -155,7 +153,7 @@ class DeepgramTranscriber(BaseTranscriber):
return data["duration"]
async def process(self, warmup=True):
extra_headers = {"Authorization": f"Token {DEEPGRAM_API_KEY}"}
extra_headers = {"Authorization": f"Token {self.api_key}"}
self.audio_queue = asyncio.Queue()
async with websockets.connect(