atom-audio-engine 0.1.2__tar.gz → 0.1.5__tar.gz

{atom_audio_engine-0.1.2 → atom_audio_engine-0.1.5}/PKG-INFO

@@ -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-0.1.2 → atom_audio_engine-0.1.5}/atom_audio_engine.egg-info/PKG-INFO

@@ -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/atom_audio_engine.egg-info/SOURCES.txt

@@ -0,0 +1,36 @@
+README.md
+pyproject.toml
+setup.py
+atom_audio_engine.egg-info/PKG-INFO
+atom_audio_engine.egg-info/SOURCES.txt
+atom_audio_engine.egg-info/dependency_links.txt
+atom_audio_engine.egg-info/requires.txt
+atom_audio_engine.egg-info/top_level.txt
+audio_engine/__init__.py
+audio_engine/asr/__init__.py
+audio_engine/asr/base.py
+audio_engine/asr/cartesia.py
+audio_engine/asr/deepgram.py
+audio_engine/core/__init__.py
+audio_engine/core/config.py
+audio_engine/core/pipeline.py
+audio_engine/core/types.py
+audio_engine/integrations/__init__.py
+audio_engine/integrations/geneface.py
+audio_engine/llm/__init__.py
+audio_engine/llm/base.py
+audio_engine/llm/groq.py
+audio_engine/pipelines/__init__.py
+audio_engine/pipelines/personaplex/__init__.py
+audio_engine/pipelines/personaplex/client.py
+audio_engine/pipelines/personaplex/config.py
+audio_engine/pipelines/personaplex/pipeline.py
+audio_engine/pipelines/personaplex/types.py
+audio_engine/pipelines/personaplex/utils.py
+audio_engine/streaming/__init__.py
+audio_engine/streaming/websocket_server.py
+audio_engine/tts/__init__.py
+audio_engine/tts/base.py
+audio_engine/tts/cartesia.py
+audio_engine/utils/__init__.py
+audio_engine/utils/audio.py

{atom_audio_engine-0.1.2 → atom_audio_engine-0.1.5}/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

{atom_audio_engine-0.1.2 → atom_audio_engine-0.1.5}/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"
         )

{atom_audio_engine-0.1.2 → atom_audio_engine-0.1.5}/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):

{atom_audio_engine-0.1.2 → atom_audio_engine-0.1.5}/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", "")

{atom_audio_engine-0.1.2 → atom_audio_engine-0.1.5}/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__)

{atom_audio_engine-0.1.2 → atom_audio_engine-0.1.5}/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",

{atom_audio_engine-0.1.2 → atom_audio_engine-0.1.5}/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)

{atom_audio_engine-0.1.2 → atom_audio_engine-0.1.5}/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:

{atom_audio_engine-0.1.2 → atom_audio_engine-0.1.5}/audio_engine/integrations/__init__.py

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

{atom_audio_engine-0.1.2 → atom_audio_engine-0.1.5}/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__)
 

{atom_audio_engine-0.1.2 → atom_audio_engine-0.1.5}/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")

{atom_audio_engine-0.1.2 → atom_audio_engine-0.1.5}/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.
 

{atom_audio_engine-0.1.2 → atom_audio_engine-0.1.5}/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.
 

{atom_audio_engine-0.1.2 → atom_audio_engine-0.1.5}/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"]

{atom_audio_engine-0.1.2 → atom_audio_engine-0.1.5}/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

{atom_audio_engine-0.1.2 → atom_audio_engine-0.1.5}/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")

{atom_audio_engine-0.1.2 → atom_audio_engine-0.1.5}/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 ""

{atom_audio_engine-0.1.2 → atom_audio_engine-0.1.5}/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:

{atom_audio_engine-0.1.2 → atom_audio_engine-0.1.5}/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,

{atom_audio_engine-0.1.2 → atom_audio_engine-0.1.5}/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 → atom_audio_engine-0.1.5}/pyproject.toml

@@ -4,7 +4,7 @@ build-backend = "setuptools.build_meta"
 
 [project]
 name = "atom-audio-engine"
-version = "0.1.2"
+version = "0.1.5"
 description = "A pluggable, async-first Python framework for real-time audio-to-audio conversational AI"
 readme = "README.md"
 requires-python = ">=3.9"
@@ -90,11 +90,6 @@ packages = [
     "audio_engine.utils",
     "audio_engine.pipelines",
     "audio_engine.pipelines.personaplex",
-    "audio_engine.examples",
-    "audio_engine.scripts",
-    "audio_engine.tests",
-    "audio_engine.tests.test_personaplex",
-    "audio_engine.tests.test_traditional_pipeline",
 ]
 
 [tool.setuptools.package-data]

atom_audio_engine-0.1.2/atom_audio_engine.egg-info/SOURCES.txt

@@ -1,61 +0,0 @@
-README.md
-pyproject.toml
-setup.py
-atom_audio_engine.egg-info/PKG-INFO
-atom_audio_engine.egg-info/SOURCES.txt
-atom_audio_engine.egg-info/dependency_links.txt
-atom_audio_engine.egg-info/requires.txt
-atom_audio_engine.egg-info/top_level.txt
-audio_engine/__init__.py
-audio_engine/asr/__init__.py
-audio_engine/asr/base.py
-audio_engine/asr/cartesia.py
-audio_engine/asr/deepgram.py
-audio_engine/core/__init__.py
-audio_engine/core/config.py
-audio_engine/core/pipeline.py
-audio_engine/core/types.py
-audio_engine/examples/__init__.py
-audio_engine/examples/basic_stt_llm_tts.py
-audio_engine/examples/geneface_animation.py
-audio_engine/examples/personaplex_pipeline.py
-audio_engine/examples/websocket_server.py
-audio_engine/integrations/__init__.py
-audio_engine/integrations/geneface.py
-audio_engine/llm/__init__.py
-audio_engine/llm/base.py
-audio_engine/llm/groq.py
-audio_engine/pipelines/__init__.py
-audio_engine/pipelines/personaplex/__init__.py
-audio_engine/pipelines/personaplex/client.py
-audio_engine/pipelines/personaplex/config.py
-audio_engine/pipelines/personaplex/pipeline.py
-audio_engine/pipelines/personaplex/types.py
-audio_engine/pipelines/personaplex/utils.py
-audio_engine/scripts/debug_pipeline.py
-audio_engine/scripts/debug_tts.py
-audio_engine/scripts/test_cartesia_connect.py
-audio_engine/streaming/__init__.py
-audio_engine/streaming/websocket_server.py
-audio_engine/tests/__init__.py
-audio_engine/tests/test_personaplex/__init__.py
-audio_engine/tests/test_personaplex/test_personaplex.py
-audio_engine/tests/test_personaplex/test_personaplex_client.py
-audio_engine/tests/test_personaplex/test_personaplex_config.py
-audio_engine/tests/test_personaplex/test_personaplex_message.py
-audio_engine/tests/test_personaplex/test_personaplex_pipeline.py
-audio_engine/tests/test_personaplex/test_personaplex_session.py
-audio_engine/tests/test_personaplex/test_personaplex_transcript.py
-audio_engine/tests/test_traditional_pipeline/__init__.py
-audio_engine/tests/test_traditional_pipeline/test_cartesia_asr.py
-audio_engine/tests/test_traditional_pipeline/test_config_env.py
-audio_engine/tests/test_traditional_pipeline/test_conversation_context.py
-audio_engine/tests/test_traditional_pipeline/test_pipeline_creation.py
-audio_engine/tests/test_traditional_pipeline/test_pipeline_with_mocks.py
-audio_engine/tests/test_traditional_pipeline/test_provider_factories.py
-audio_engine/tests/test_traditional_pipeline/test_websocket_server.py
-audio_engine/tts/__init__.py
-audio_engine/tts/base.py
-audio_engine/tts/cartesia.py
-audio_engine/utils/__init__.py
-audio_engine/utils/audio.py

atom_audio_engine-0.1.2/audio_engine/examples/__init__.py

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