Use Parameters instead of Attributes in docstrings to fix duplicate object warnings

Napoleon's Attributes section creates class-level attribute docs that
duplicate the __init__ parameter docs when napoleon_include_init_with_doc
is enabled. Using Parameters avoids the duplication.
This commit is contained in:
Mark Backman
2026-04-03 10:36:36 -04:00
parent f078b8b867
commit c2358b273b
14 changed files with 16 additions and 16 deletions

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@@ -40,7 +40,7 @@ class TranscriptHandler:
Maintains a list of conversation messages and outputs them either to a log
or to a file as they are received. Each message includes its timestamp and role.
Attributes:
Parameters:
messages: List of all processed transcript messages
output_file: Optional path to file where transcript is saved. If None, outputs to log only.
"""

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@@ -27,7 +27,7 @@ from pipecat.services.xai.realtime import events
class GrokRealtimeLLMInvocationParams(TypedDict):
"""Context-based parameters for invoking Grok Realtime API.
Attributes:
Parameters:
system_instruction: System prompt/instructions for the session.
messages: List of conversation items formatted for Grok Realtime.
tools: List of tool definitions (function, web_search, x_search, file_search).

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@@ -36,7 +36,7 @@ class FrameOrder(Enum):
When multiple parallel pipelines produce output for the same input frame,
this setting determines the order in which those output frames are pushed.
Attributes:
Parameters:
ARRIVAL: Frames are pushed in the order they arrive from any pipeline.
This is the default and matches the behavior of prior versions.
PIPELINE: Frames are pushed in pipeline definition order — all frames

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@@ -123,7 +123,7 @@ class GenesysAudioHookSerializer(FrameSerializer):
# Set output variables to return to Architect
serializer.set_output_variables({"intent": "billing", "resolved": True})
Attributes:
Parameters:
PROTOCOL_VERSION: The AudioHook protocol version (currently "2").
"""
@@ -132,7 +132,7 @@ class GenesysAudioHookSerializer(FrameSerializer):
class InputParams(FrameSerializer.InputParams):
"""Configuration parameters for GenesysAudioHookSerializer.
Attributes:
Parameters:
genesys_sample_rate: Sample rate used by Genesys (default: 8000 Hz).
sample_rate: Optional override for pipeline input sample rate.
channel: Which audio channels to process (external, internal, both).

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@@ -83,7 +83,7 @@ class HeyGenClient:
1. WebSocket connection for avatar control and audio streaming
2. LiveKit connection for receiving avatar video and audio
Attributes:
Parameters:
HEY_GEN_SAMPLE_RATE (int): The required sample rate for HeyGen's audio processing (24000 Hz)
"""

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@@ -86,7 +86,7 @@ def language_to_sarvam_language(language: Language) -> str:
class ModelConfig:
"""Immutable configuration for a Sarvam STT model.
Attributes:
Parameters:
supports_prompt: Whether the model accepts prompt parameter.
supports_mode: Whether the model accepts mode parameter.
supports_language: Whether the model accepts language parameter.

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@@ -75,7 +75,7 @@ except ModuleNotFoundError as e:
class SarvamTTSModel(str, Enum):
"""Available Sarvam TTS models.
Attributes:
Parameters:
BULBUL_V2: Standard TTS model with pitch/loudness control.
- Supports pitch, loudness, pace (0.3-3.0)
- Default sample rate: 22050 Hz
@@ -145,7 +145,7 @@ class SarvamTTSSpeakerV3(str, Enum):
class TTSModelConfig:
"""Immutable configuration for a Sarvam TTS model.
Attributes:
Parameters:
supports_pitch: Whether the model accepts pitch parameter.
supports_loudness: Whether the model accepts loudness parameter.
supports_temperature: Whether the model accepts temperature parameter.

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@@ -67,7 +67,7 @@ from pipecat.utils.time import seconds_to_nanoseconds
class TTSContext:
"""Context information for a TTS request.
Attributes:
Parameters:
append_to_context: Whether this TTS output should be appended to the
conversation context after it is spoken.
push_assistant_aggregation: Whether to push an

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@@ -37,7 +37,7 @@ class WhatsAppClient:
events from WhatsApp, and maintains ongoing call state. It supports both
incoming call handling and call termination through the WhatsApp Cloud API.
Attributes:
Parameters:
_whatsapp_api: WhatsApp API instance for making API calls
_ongoing_calls_map: Dictionary mapping call IDs to WebRTC connections
_ice_servers: List of ICE servers for WebRTC connections

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@@ -15,7 +15,7 @@ class ProcessFrameResult(Enum):
Controls whether the strategy loop in the controller continues to the
next strategy or stops early.
Attributes:
Parameters:
CONTINUE: Continue to the next strategy in the loop.
STOP: Stop evaluating further strategies for this frame.
"""

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@@ -24,7 +24,7 @@ class UserTurnStartedParams:
contextual information about how the user turn should be handled by the user
aggregator.
Attributes:
Parameters:
enable_user_speaking_frames: Whether the user aggregator should emit
frames indicating user speaking state (e.g., user started speaking)
during the bot's turn. This is typically enabled by default, but may

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@@ -24,7 +24,7 @@ class UserTurnStoppedParams:
contextual information about how the end of user turn should be handled by
the user aggregator.
Attributes:
Parameters:
enable_user_speaking_frames: Whether the user aggregator should emit
frames indicating user speaking state (e.g., user stopped speaking).
This is typically enabled by default, but may be disabled when another

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@@ -147,7 +147,7 @@ Remember: Focus on conversational completeness and how long the user might need.
class UserTurnCompletionConfig:
"""Configuration for turn completion behavior.
Attributes:
Parameters:
instructions: Custom instructions for turn completion. If not provided,
uses default USER_TURN_COMPLETION_INSTRUCTIONS.
incomplete_short_timeout: Seconds to wait after short incomplete (○) before prompting.

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@@ -57,7 +57,7 @@ class UserTurnStrategies:
start: [VADUserTurnStartStrategy, TranscriptionUserTurnStartStrategy]
stop: [TurnAnalyzerUserTurnStopStrategy(LocalSmartTurnAnalyzerV3)]
Attributes:
Parameters:
start: A list of user turn start strategies used to detect when
the user starts speaking.
stop: A list of user turn stop strategies used to decide when