atom-audio-engine 0.1.2__py3-none-any.whl → 0.1.5__py3-none-any.whl

{atom_audio_engine-0.1.2.dist-info → atom_audio_engine-0.1.5.dist-info}/METADATA

@@ -1,6 +1,6 @@
 Metadata-Version: 2.4
 Name: atom-audio-engine
-Version: 0.1.2
+Version: 0.1.5
 Summary: A pluggable, async-first Python framework for real-time audio-to-audio conversational AI
 Author-email: ATOM Group <info@atomgroup.ng>
 License-Expression: MIT

atom_audio_engine-0.1.5.dist-info/RECORD

@@ -0,0 +1,32 @@
+audio_engine/__init__.py,sha256=6AuBeSWgRhAUEMIuIpPkJaATDjB5pttdUDujWtvC-8Q,1499
+audio_engine/asr/__init__.py,sha256=VlQ09fZf__4Dn2EGV3RnsOvwrKMPBemDRKNTHHnYFHY,1193
+audio_engine/asr/base.py,sha256=dC6cvAlxoOzUXNz9do-ueRObmNXqg4rtsY9fh6k67vA,2380
+audio_engine/asr/cartesia.py,sha256=jLoetv6jiAbcfmfGkrwKRGo5sbMZ-WJTwIVvWno1Vko,13185
+audio_engine/asr/deepgram.py,sha256=RwtG0e66Y_8HKehoHFzjgJ5JlQs8JEpu_0kpWGmsvBc,6431
+audio_engine/core/__init__.py,sha256=aqN1HeFZGX0xAnVyfURdExXXxxmwwZEq_KFj6MqXZwo,289
+audio_engine/core/config.py,sha256=Jmp-rr0MXXejF3qtx1-CPCEwQOorggHIH5cV3t8G-I4,5205
+audio_engine/core/pipeline.py,sha256=rMZOlllT32xruz3nkeoYGRfnq94zgs-dzAbTahSITtU,8808
+audio_engine/core/types.py,sha256=iFQPajgeS1YgMWXJvubA8sWbxLI1Z8nF-z1uucrgNm4,2295
+audio_engine/integrations/__init__.py,sha256=69Hna1pfmB929WbM7GpAHlrk4xPOleKTnoaBBksFo9k,114
+audio_engine/integrations/geneface.py,sha256=JgxGYfqDk9n-p4e1VNczoEJdMPzzfF5QGsyxxinrWr8,8790
+audio_engine/llm/__init__.py,sha256=kJpCYylJMsbO_oA5XFcLKBI2N-ZCOVdwUduU_jbh3po,977
+audio_engine/llm/base.py,sha256=vsKi2UYuhMr_nubMsoyU6hzSV6gr3DZ1sPvqkSvap3c,2862
+audio_engine/llm/groq.py,sha256=zX4z_ZPyB5_FxhMwg-MnK5Ga6vpqRJLaRTrknSYJZXU,6682
+audio_engine/pipelines/__init__.py,sha256=Q1iZjX38TigrZPBaFgv_5AXw21wBN1Z-4nfXPjV-xDI,49
+audio_engine/pipelines/personaplex/__init__.py,sha256=nX37MS93pYUPKiYwY2aa9G-PEI4x2yKjdLqGeab7wWI,916
+audio_engine/pipelines/personaplex/client.py,sha256=NAiG6V9nTWh8ozrb5jT-6h8fesTuJZDgh-l7DlHQm6M,8667
+audio_engine/pipelines/personaplex/config.py,sha256=6fBteI-HjJJl3ZcK5QZCCa9kcKVNDgPptLIkJNZc9kg,2935
+audio_engine/pipelines/personaplex/pipeline.py,sha256=WUkFalPQ9sxICeFpF-58HJxzfQ30vfZ4WAs-E5aI60s,10411
+audio_engine/pipelines/personaplex/types.py,sha256=6MvU2hBukBflJxat3MtC6bGQY1b33jaOIiOi2tZJRnU,4727
+audio_engine/pipelines/personaplex/utils.py,sha256=um_7nGRFH0QaLIIfLwPnBXgFW0fVGU7gkjF8Gm-Hq4U,5000
+audio_engine/streaming/__init__.py,sha256=0FOorloUtads4ZeJKLdlTcqaL0l2G7Byq4ijQG3W1Fk,127
+audio_engine/streaming/websocket_server.py,sha256=p9_ugvXfUW0TeuCkTCTUoHCmQ0vBTGq2J2Ubys1HdeY,10896
+audio_engine/tts/__init__.py,sha256=sJIN_PgwO1_GVtr6NBR8oegqzLa9oR_qqD6ixbkDVro,967
+audio_engine/tts/base.py,sha256=YbvdVF9XbJyv7NWf_5W7bawyquz8z83BcNcXOdA6iaY,4424
+audio_engine/tts/cartesia.py,sha256=VipeNLgOac_hlsi2hasZe_ALYZZ7hvQ72eFeY17J600,16313
+audio_engine/utils/__init__.py,sha256=J-XxXjgjAmvsM39W3pKI2we-C6S7rd49zfaEf9omwN8,245
+audio_engine/utils/audio.py,sha256=RHp-FRjyCMPDaSQCOnxp7m_KO2z3Enu3iX7J5BVRD-0,5507
+atom_audio_engine-0.1.5.dist-info/METADATA,sha256=4326-a-wzhsbhZW-O4KRUetl6Y_Js8GcH1Mu4_7iXi4,6690
+atom_audio_engine-0.1.5.dist-info/WHEEL,sha256=wUyA8OaulRlbfwMtmQsvNngGrxQHAvkKcvRmdizlJi0,92
+atom_audio_engine-0.1.5.dist-info/top_level.txt,sha256=IyumwgFrsDL7nlZlBijX-0shiSVhhBCFPUNBRNKzWP4,13
+atom_audio_engine-0.1.5.dist-info/RECORD,,

audio_engine/__init__.py

@@ -4,7 +4,7 @@ Audio Engine - Pluggable audio-to-audio conversational AI framework.
 Orchestrates ASR → LLM → TTS pipeline with real-time streaming support.
 """
 
-__version__ = "0.1.0"
+__version__ = "0.1.5"
 
 # Core exports
 from .core.pipeline import Pipeline

audio_engine/asr/__init__.py

@@ -1,6 +1,6 @@
 """ASR (Speech-to-Text) providers."""
 
-from core.config import ASRConfig
+from ..core.config import ASRConfig
 
 from .base import BaseASR
 from .deepgram import DeepgramASR
@@ -40,6 +40,5 @@ def get_asr_from_config(config: ASRConfig) -> BaseASR:
         )
     else:
         raise ValueError(
-            f"Unknown ASR provider: {config.provider}. "
-            f"Supported: deepgram, cartesia"
+            f"Unknown ASR provider: {config.provider}. " f"Supported: deepgram, cartesia"
         )

audio_engine/asr/base.py

@@ -3,7 +3,7 @@
 from abc import ABC, abstractmethod
 from typing import AsyncIterator, Optional
 
-from core.types import AudioChunk, TranscriptChunk
+from ..core.types import AudioChunk, TranscriptChunk
 
 
 class BaseASR(ABC):

audio_engine/asr/cartesia.py

@@ -8,7 +8,7 @@ from urllib.parse import quote
 
 import websockets
 
-from core.types import AudioChunk, TranscriptChunk
+from ..core.types import AudioChunk, TranscriptChunk
 from .base import BaseASR
 
 logger = logging.getLogger(__name__)
@@ -135,9 +135,7 @@ class CartesiaASR(BaseASR):
                 logger.debug("Cartesia WebSocket connected")
             except asyncio.TimeoutError:
                 logger.error(f"WebSocket connection timeout to {url}")
-                raise TimeoutError(
-                    "Failed to connect to Cartesia WebSocket within 30s timeout"
-                )
+                raise TimeoutError("Failed to connect to Cartesia WebSocket within 30s timeout")
 
             # Start background receive task
             self._receive_task = asyncio.create_task(self._receive_loop())
@@ -232,9 +230,7 @@ class CartesiaASR(BaseASR):
             transcript_parts = []
             while True:
                 try:
-                    response = await asyncio.wait_for(
-                        self._response_queue.get(), timeout=10.0
-                    )
+                    response = await asyncio.wait_for(self._response_queue.get(), timeout=10.0)
 
                     if response.get("type") == "transcript":
                         text = response.get("text", "")
@@ -320,9 +316,7 @@ class CartesiaASR(BaseASR):
             if done_sent:
                 while True:
                     try:
-                        response = await asyncio.wait_for(
-                            self._response_queue.get(), timeout=5.0
-                        )
+                        response = await asyncio.wait_for(self._response_queue.get(), timeout=5.0)
 
                         if response.get("type") == "transcript":
                             text = response.get("text", "")

audio_engine/asr/deepgram.py

@@ -5,7 +5,7 @@ from typing import AsyncIterator, Optional
 
 from deepgram import DeepgramClient
 
-from core.types import AudioChunk, TranscriptChunk
+from ..core.types import AudioChunk, TranscriptChunk
 from .base import BaseASR
 
 logger = logging.getLogger(__name__)

audio_engine/core/__init__.py

@@ -1,8 +1,8 @@
 """Core pipeline and configuration."""
 
-from core.pipeline import Pipeline
-from core.config import AudioEngineConfig
-from core.types import AudioChunk, TranscriptChunk, ResponseChunk
+from .pipeline import Pipeline
+from .config import AudioEngineConfig
+from .types import AudioChunk, TranscriptChunk, ResponseChunk
 
 __all__ = [
     "Pipeline",

audio_engine/core/config.py

@@ -144,10 +144,10 @@ class AudioEngineConfig:
         Raises:
             ValueError: If provider initialization fails
         """
-        from asr import get_asr_from_config
-        from llm import get_llm_from_config
-        from tts import get_tts_from_config
-        from core.pipeline import Pipeline
+        from ..asr import get_asr_from_config
+        from ..llm import get_llm_from_config
+        from ..tts import get_tts_from_config
+        from .pipeline import Pipeline
 
         asr = get_asr_from_config(self.asr)
         llm = get_llm_from_config(self.llm)

audio_engine/core/pipeline.py

@@ -5,10 +5,10 @@ import logging
 import time
 from typing import AsyncIterator, Optional, Callable, Any
 
-from asr.base import BaseASR
-from llm.base import BaseLLM
-from tts.base import BaseTTS
-from core.types import (
+from ..asr.base import BaseASR
+from ..llm.base import BaseLLM
+from ..tts.base import BaseTTS
+from .types import (
     AudioChunk,
     TranscriptChunk,
     ResponseChunk,
@@ -165,9 +165,7 @@ class Pipeline:
 
         return response_audio
 
-    async def stream(
-        self, audio_stream: AsyncIterator[AudioChunk]
-    ) -> AsyncIterator[AudioChunk]:
+    async def stream(self, audio_stream: AsyncIterator[AudioChunk]) -> AsyncIterator[AudioChunk]:
         """
         Process streaming audio input and yield streaming audio output.
 
@@ -208,9 +206,7 @@ class Pipeline:
 
         async def llm_text_stream() -> AsyncIterator[str]:
             nonlocal response_buffer
-            async for chunk in self.llm.generate_stream(
-                transcript_buffer, self.context
-            ):
+            async for chunk in self.llm.generate_stream(transcript_buffer, self.context):
                 response_buffer += chunk.text
                 yield chunk.text
                 if chunk.is_final:

audio_engine/integrations/__init__.py

@@ -1,5 +1,5 @@
 """External system integrations."""
 
-from integrations.geneface import GeneFaceIntegration
+from .geneface import GeneFaceIntegration
 
 __all__ = ["GeneFaceIntegration"]

audio_engine/integrations/geneface.py

@@ -8,7 +8,7 @@ from pathlib import Path
 from typing import Optional, AsyncIterator
 from dataclasses import dataclass
 
-from core.types import AudioChunk
+from ..core.types import AudioChunk
 
 logger = logging.getLogger(__name__)
 

audio_engine/llm/__init__.py

@@ -1,6 +1,6 @@
 """LLM (Large Language Model) providers."""
 
-from core.config import LLMConfig
+from ..core.config import LLMConfig
 
 from .base import BaseLLM
 from .groq import GroqLLM
@@ -33,6 +33,4 @@ def get_llm_from_config(config: LLMConfig) -> BaseLLM:
             **config.extra,
         )
     else:
-        raise ValueError(
-            f"Unknown LLM provider: {config.provider}. " f"Supported: groq"
-        )
+        raise ValueError(f"Unknown LLM provider: {config.provider}. " f"Supported: groq")

audio_engine/llm/base.py

@@ -3,7 +3,7 @@
 from abc import ABC, abstractmethod
 from typing import AsyncIterator, Optional
 
-from core.types import ResponseChunk, ConversationContext
+from ..core.types import ResponseChunk, ConversationContext
 
 
 class BaseLLM(ABC):
@@ -21,7 +21,7 @@ class BaseLLM(ABC):
         temperature: float = 0.7,
         max_tokens: int = 1024,
         system_prompt: Optional[str] = None,
-        **kwargs
+        **kwargs,
     ):
         """
         Initialize the LLM provider.
@@ -42,9 +42,7 @@ class BaseLLM(ABC):
         self.config = kwargs
 
     @abstractmethod
-    async def generate(
-        self, prompt: str, context: Optional[ConversationContext] = None
-    ) -> str:
+    async def generate(self, prompt: str, context: Optional[ConversationContext] = None) -> str:
         """
         Generate a complete response to a prompt.
 

audio_engine/llm/groq.py

@@ -5,7 +5,7 @@ from typing import AsyncIterator, Optional
 
 from groq import Groq
 
-from core.types import ResponseChunk, ConversationContext
+from ..core.types import ResponseChunk, ConversationContext
 from .base import BaseLLM
 
 logger = logging.getLogger(__name__)
@@ -85,9 +85,7 @@ class GroqLLM(BaseLLM):
             except Exception as e:
                 logger.error(f"Error disconnecting Groq: {e}")
 
-    async def generate(
-        self, prompt: str, context: Optional[ConversationContext] = None
-    ) -> str:
+    async def generate(self, prompt: str, context: Optional[ConversationContext] = None) -> str:
         """
         Generate a complete response to a prompt.
 

audio_engine/streaming/__init__.py

@@ -1,5 +1,5 @@
 """Streaming and WebSocket server components."""
 
-from streaming.websocket_server import WebSocketServer
+from .websocket_server import WebSocketServer
 
 __all__ = ["WebSocketServer"]

audio_engine/streaming/websocket_server.py

@@ -7,9 +7,9 @@ from typing import Optional, Callable, Any
 
 import websockets
 
-from core.pipeline import Pipeline
-from core.types import AudioChunk, AudioFormat
-from core.config import AudioEngineConfig
+from ..core.pipeline import Pipeline
+from ..core.types import AudioChunk, AudioFormat
+from ..core.config import AudioEngineConfig
 
 logger = logging.getLogger(__name__)
 
@@ -62,9 +62,7 @@ class WebSocketServer:
             on_disconnect: Callback when client disconnects
         """
         if websockets is None:
-            raise ImportError(
-                "websockets package required. Install with: pip install websockets"
-            )
+            raise ImportError("websockets package required. Install with: pip install websockets")
 
         self.pipeline = pipeline
         self.host = host
@@ -137,9 +135,7 @@ class WebSocketServer:
             if self.on_disconnect:
                 self.on_disconnect(client_id)
 
-    async def _process_client_stream(
-        self, websocket: WebSocketServerProtocol, client_id: str
-    ):
+    async def _process_client_stream(self, websocket: WebSocketServerProtocol, client_id: str):
         """Process streaming audio from a client."""
         audio_queue: asyncio.Queue[AudioChunk] = asyncio.Queue()
         end_of_speech = asyncio.Event()
@@ -231,12 +227,8 @@ class WebSocketServer:
                     original_on_llm_response(text)
 
             # Temporarily override callbacks
-            self.pipeline.on_transcript = lambda t: asyncio.create_task(
-                send_transcript(t)
-            )
-            self.pipeline.on_llm_response = lambda t: asyncio.create_task(
-                send_llm_response(t)
-            )
+            self.pipeline.on_transcript = lambda t: asyncio.create_task(send_transcript(t))
+            self.pipeline.on_llm_response = lambda t: asyncio.create_task(send_llm_response(t))
 
             try:
                 # Wait for some audio to arrive

audio_engine/tts/__init__.py

@@ -1,6 +1,6 @@
 """TTS (Text-to-Speech) providers."""
 
-from core.config import TTSConfig
+from ..core.config import TTSConfig
 
 from .base import BaseTTS
 from .cartesia import CartesiaTTS
@@ -32,6 +32,4 @@ def get_tts_from_config(config: TTSConfig) -> BaseTTS:
             **config.extra,
         )
     else:
-        raise ValueError(
-            f"Unknown TTS provider: {config.provider}. " f"Supported: cartesia"
-        )
+        raise ValueError(f"Unknown TTS provider: {config.provider}. " f"Supported: cartesia")

audio_engine/tts/base.py

@@ -3,7 +3,7 @@
 from abc import ABC, abstractmethod
 from typing import AsyncIterator, Optional
 
-from core.types import AudioChunk, AudioFormat
+from ..core.types import AudioChunk, AudioFormat
 
 
 class BaseTTS(ABC):
@@ -21,7 +21,7 @@ class BaseTTS(ABC):
         model: Optional[str] = None,
         speed: float = 1.0,
         output_format: AudioFormat = AudioFormat.PCM_24K,
-        **kwargs
+        **kwargs,
     ):
         """
         Initialize the TTS provider.
@@ -97,9 +97,7 @@ class BaseTTS(ABC):
                     sentence = parts[0] + ender
 
                     if sentence.strip():
-                        async for audio_chunk in self.synthesize_stream(
-                            sentence.strip()
-                        ):
+                        async for audio_chunk in self.synthesize_stream(sentence.strip()):
                             yield audio_chunk
 
                     buffer = parts[1] if len(parts) > 1 else ""

audio_engine/tts/cartesia.py

@@ -9,7 +9,7 @@ from typing import AsyncIterator, Optional
 import websockets
 
 
-from core.types import AudioChunk, AudioFormat
+from ..core.types import AudioChunk, AudioFormat
 from .base import BaseTTS
 
 logger = logging.getLogger(__name__)
@@ -138,9 +138,7 @@ class CartesiaTTS(BaseTTS):
             AudioChunk objects with audio data
         """
         if websockets is None:
-            raise ImportError(
-                "websockets package required. Install: pip install websockets"
-            )
+            raise ImportError("websockets package required. Install: pip install websockets")
 
         if not self.api_key:
             raise ValueError("api_key required for Cartesia TTS")
@@ -151,16 +149,12 @@ class CartesiaTTS(BaseTTS):
         context_id = str(uuid.uuid4())
 
         ws_url = (
-            f"{self.WS_URL}"
-            f"?api_key={self.api_key}"
-            f"&cartesia_version={self.CARTESIA_VERSION}"
+            f"{self.WS_URL}" f"?api_key={self.api_key}" f"&cartesia_version={self.CARTESIA_VERSION}"
         )
 
         try:
             async with websockets.connect(ws_url) as websocket:
-                logger.debug(
-                    f"Cartesia TTS WebSocket connected | Context: {context_id}"
-                )
+                logger.debug(f"Cartesia TTS WebSocket connected | Context: {context_id}")
 
                 # Task to receive audio from WebSocket
                 async def receive_audio():
@@ -175,9 +169,7 @@ class CartesiaTTS(BaseTTS):
                                         f"Cartesia: received response type={response.get('type')}"
                                     )
                                     # Handle audio chunk (base64 in "data" field)
-                                    if response.get("type") == "chunk" and response.get(
-                                        "data"
-                                    ):
+                                    if response.get("type") == "chunk" and response.get("data"):
                                         audio_bytes = base64.b64decode(response["data"])
                                         yield audio_bytes
                                         logger.debug(
@@ -199,17 +191,13 @@ class CartesiaTTS(BaseTTS):
                                             or str(response)
                                         )
                                         logger.error(f"Cartesia TTS error: {error_msg}")
-                                        raise RuntimeError(
-                                            f"Cartesia API error: {error_msg}"
-                                        )
+                                        raise RuntimeError(f"Cartesia API error: {error_msg}")
                                     else:
                                         logger.debug(
                                             f"Cartesia: response type {response.get('type')}"
                                         )
                                 except json.JSONDecodeError:
-                                    logger.warning(
-                                        f"Failed to parse Cartesia response: {message}"
-                                    )
+                                    logger.warning(f"Failed to parse Cartesia response: {message}")
                     except Exception as e:
                         logger.error(f"Cartesia receive error: {e}", exc_info=True)
                         raise
@@ -260,9 +248,7 @@ class CartesiaTTS(BaseTTS):
                                         },
                                     }
                                     await websocket.send(json.dumps(request))
-                                    logger.debug(
-                                        f"Cartesia: sent text on timeout (continue=true)"
-                                    )
+                                    logger.debug(f"Cartesia: sent text on timeout (continue=true)")
                                     accumulated_text = ""
                                 continue
 
@@ -287,9 +273,7 @@ class CartesiaTTS(BaseTTS):
                                         },
                                     }
                                     await websocket.send(json.dumps(request))
-                                    logger.debug(
-                                        f"Cartesia: sent final text (continue=false)"
-                                    )
+                                    logger.debug(f"Cartesia: sent final text (continue=false)")
                                 else:
                                     # Send empty transcript to signal end
                                     request = {
@@ -309,9 +293,7 @@ class CartesiaTTS(BaseTTS):
                                         },
                                     }
                                     await websocket.send(json.dumps(request))
-                                    logger.debug(
-                                        "Cartesia: sent empty transcript to signal end"
-                                    )
+                                    logger.debug("Cartesia: sent empty transcript to signal end")
                                 logger.info("Cartesia: all text sent")
                                 break
 
@@ -322,9 +304,7 @@ class CartesiaTTS(BaseTTS):
                             )
 
                             # Send when buffer is large enough or ends with punctuation
-                            if len(accumulated_text) > 30 or token.endswith(
-                                (".", "!", "?")
-                            ):
+                            if len(accumulated_text) > 30 or token.endswith((".", "!", "?")):
                                 request = {
                                     "model_id": self.model,
                                     "transcript": accumulated_text,
@@ -342,9 +322,7 @@ class CartesiaTTS(BaseTTS):
                                     },
                                 }
                                 await websocket.send(json.dumps(request))
-                                logger.debug(
-                                    f"Cartesia: sent buffered text (continue=true)"
-                                )
+                                logger.debug(f"Cartesia: sent buffered text (continue=true)")
                                 accumulated_text = ""
 
                     except Exception as e:

audio_engine/utils/__init__.py

@@ -1,6 +1,6 @@
 """Utility functions for the audio engine."""
 
-from utils.audio import (
+from .audio import (
     resample_audio,
     pcm_to_wav,
     wav_to_pcm,

audio_engine/utils/audio.py

@@ -58,9 +58,7 @@ def _simple_resample(
     """Simple linear interpolation resampling."""
     if sample_width == 2:
         fmt = "<h"
-        samples = [
-            struct.unpack(fmt, audio[i : i + 2])[0] for i in range(0, len(audio), 2)
-        ]
+        samples = [struct.unpack(fmt, audio[i : i + 2])[0] for i in range(0, len(audio), 2)]
     else:
         raise ValueError(f"Unsupported sample width: {sample_width}")
 

atom_audio_engine-0.1.2.dist-info/RECORD

@@ -1,57 +0,0 @@
-audio_engine/__init__.py,sha256=AQ0uto-Jn3cNqW35MMtSyX5mhXJMFv9AQhjcAkqZ7L4,1499
-audio_engine/asr/__init__.py,sha256=w0t2ahxgApZbZjSc748tN3tmKDeXzasfBh51ZjPF9uc,1203
-audio_engine/asr/base.py,sha256=MFC_7HmyEDnhDwUn62CWZsiF9_-mBVVsUK-Yppiq4Vk,2378
-audio_engine/asr/cartesia.py,sha256=BXnvscO9VaR3LsfEGn7lJ66udzUjz44JzZTmSizZqIg,13321
-audio_engine/asr/deepgram.py,sha256=M59lgrVFMS6-3YQcYaUY7cUdt2-MBptt_VExdfnSXr0,6429
-audio_engine/core/__init__.py,sha256=7naTEkqDjrPsejviXk662OR86xVCyckU7eMKVpjwYys,301
-audio_engine/core/config.py,sha256=EF98O2Gt8q29FX3T6UeDwWNIbm77bni99SThiJKl5Tk,5203
-audio_engine/core/pipeline.py,sha256=jX9jAlIfwU6V8GjqjivyK8Y7P41S-QS8xKYv5c9_qG0,8850
-audio_engine/core/types.py,sha256=iFQPajgeS1YgMWXJvubA8sWbxLI1Z8nF-z1uucrgNm4,2295
-audio_engine/examples/__init__.py,sha256=4oFCZaD-vg0o48hnj03ZsktG2JrtwJ7HXUYOwEYSNCY,44
-audio_engine/examples/basic_stt_llm_tts.py,sha256=tw8IIAL0WSG2M9U5SuLri75AOb7YM-twvAVAspaYVQM,6354
-audio_engine/examples/geneface_animation.py,sha256=ogjQAqPHT5EW6X3R8hn0tJwj-_QBbPiBFDZDl_olTGo,2945
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audio_engine/examples/__init__.py

@@ -1 +0,0 @@
-"""Example scripts for the audio engine."""