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Merge pull request #2265 from pipecat-ai/filipi/small_webrtc_buffer_processor

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Fixed an issue in AudioBufferProcessor when using SmallWebRTCTransport
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Filipi da Silva Fuchter
2025-07-28 09:23:58 -03:00
committed by GitHub
parent 434c3fc527 e86b55e2b3
commit 9d5510ee47
2 changed files with 5 additions and 2 deletions
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3
CHANGELOG.md
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@@ -71,6 +71,9 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
### Fixed ### Fixed
- Fixed an issue in `AudioBufferProcessor` when using `SmallWebRTCTransport` where, if
the microphone was muted, track timing was not respected.
- Fixed an issue in `AudioBufferProcessor` that caused garbled audio when - Fixed an issue in `AudioBufferProcessor` that caused garbled audio when
`enable_turn_audio` was enabled and audio resampling was required. `enable_turn_audio` was enabled and audio resampling was required.

4
src/pipecat/transports/network/webrtc_connection.py
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@@ -137,9 +137,9 @@ class SmallWebRTCTrack:
"""Receive the next frame from the track. """Receive the next frame from the track.
Returns: Returns:
The next frame if the track is enabled, None otherwise. The next frame, except for video tracks, where it returns the frame only if the track is enabled, otherwise, returns None.
""" """
if not self._enabled: if not self._enabled and self._track.kind == "video":
return None return None
return await self._track.recv() return await self._track.recv()