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Fixed an audio mixer issue when used alongside SmallWebRTCTransport.

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Filipi Fuchter
2025-05-05 17:39:18 -03:00
parent 75d261639f
commit d75815ba48
2 changed files with 11 additions and 1 deletions
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9
CHANGELOG.md
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@@ -5,6 +5,15 @@ All notable changes to **Pipecat** will be documented in this file.
The format is based on [Keep a Changelog](https://keepachangelog.com/en/1.0.0/), The format is based on [Keep a Changelog](https://keepachangelog.com/en/1.0.0/),
and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0.html). and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0.html).
## [Unreleased]
### Fixed
- Refactored how the `start` method is handled in `SmallWebRTCOutputTransport` by
initializing it before the parent class. This fixes an audio mixer issue when used
alongside `SmallWebRTCTransport`, preventing unnecessary CPU usage and avoiding the
output being flooded with silent frames when no new audio is available.
## [0.0.66] - 2025-05-02 ## [0.0.66] - 2025-05-02
### Added ### Added

3
src/pipecat/transports/network/small_webrtc.py
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@@ -484,9 +484,10 @@ class SmallWebRTCOutputTransport(BaseOutputTransport):
self._params = params self._params = params
async def start(self, frame: StartFrame): async def start(self, frame: StartFrame):
await super().start(frame)
await self._client.setup(self._params, frame) await self._client.setup(self._params, frame)
await self._client.connect() await self._client.connect()
# Parent start.
await super().start(frame)
async def stop(self, frame: EndFrame): async def stop(self, frame: EndFrame):
await super().stop(frame) await super().stop(frame)