Merge pull request #238 from pipecat-ai/aleix/pipecat-0.0.31

pipecat 0.0.31

CHANGELOG.md

@@ -5,6 +5,12 @@ 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/),
 and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0.html).
 
+## [0.0.31] - 2024-06-13
+
+### Performance
+
+- Break long audio frames into 20ms chunks instead of 10ms.
+
 ## [0.0.30] - 2024-06-13
 
 ### Added
@@ -18,19 +24,19 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
 - Allow specifying frame processors' name through a new `name` constructor
   argument.
 
+- Added `DeepgramSTTService`. This service has an ongoing websocket
+  connection. To handle this, it subclasses `AIService` instead of
+  `STTService`. The output of this service will be pushed from the same task,
+  except system frames like `StartFrame`, `CancelFrame` or
+  `StartInterruptionFrame`.
+
 ### Changed
 
 - `FrameSerializer.deserialize()` can now return `None` in case it is not
   possible to desearialize the given data.
 
 - `daily_rest.DailyRoomProperties` now allows extra unknown parameters.
 
-- Added `DeepgramSTTService`. This service has an ongoing websocket
-  connection. To handle this, it subclasses `AIService` instead of
-  `STTService`. The output of this service will be pushed from the same task,
-  except system frames like `StartFrame`, `CancelFrame` or
-  `StartInterruptionFrame`.
-
 ### Fixed
 
 - Fixed an issue where `DailyRoomProperties.exp` always had the same old

src/pipecat/transports/base_output.py

@@ -59,11 +59,11 @@ def __init__(self, params: TransportParams, **kwargs):
         self._stopped_event = asyncio.Event()
         self._is_interrupted = threading.Event()
 
-        # We will send 10ms audio which is the smallest possible amount of audio
-        # we can send. If we receive long audio frames we will chunk them into
-        # 10ms. This will help with interruption handling.
-        self._audio_chunk_size = int(self._params.audio_out_sample_rate / 100) * \
+        # We will write 20ms audio at a time. If we receive long audio frames we
+        # will chunk them. This will help with interruption handling.
+        audio_bytes_10ms = int(self._params.audio_out_sample_rate / 100) * \
             self._params.audio_out_channels * 2
+        self._audio_chunk_size = audio_bytes_10ms * 2
 
         # Create push frame task. This is the task that will push frames in
         # order. We also guarantee that all frames are pushed in the same task.