AudioBufferProcessor: record with lowest sample rate

Fixes #1653

CHANGELOG.md

@@ -37,13 +37,21 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
 
 - Upgraded `daily-python` to 0.19.3.
 
+### Deprecated
+
+- `AudioBufferProcessor` parameter `user_continuos_stream` is deprecated.
+
 ### Fixed
 
+- Fixed an `AudioBufferProcessor` issue that was causing crackling on the audio
+  stream with lower sample rate (due to upsampling the other stream). We now
+  record with the lowest sample rate to avoid upsampling.
+
 - Fixed an issue that would cause multiple `PipelineTask.on_idle_timeout`
   events to be triggered repeatedly.
 
-- Fixed an issue that was causing user and bot speech to not be synchronized
+- Fixed an `AudioBufferProcessor` issue that was causing user and bot speech to
-  during recordings.
+  not be synchronized during recordings.
 
 - Fixed an issue where voice settings weren't applied to ElevenLabsTTSService.
 
@@ -55,10 +63,6 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
 
 - Fixed an issue where `GoogleLLMService`'s TTFB value was incorrect.
 
-### Deprecated
-
-- `AudioBufferProcessor` parameter `user_continuos_stream` is deprecated.
-
 ### Other
 
 - Rename `14e-function-calling-gemini.py` to `14e-function-calling-google.py`.

src/pipecat/processors/audio/audio_buffer_processor.py

@@ -7,6 +7,8 @@
 import time
 from typing import Optional
 
+from loguru import logger
+
 from pipecat.audio.utils import create_default_resampler, interleave_stereo_audio, mix_audio
 from pipecat.frames.frames import (
     AudioRawFrame,
@@ -181,7 +183,14 @@ class AudioBufferProcessor(FrameProcessor):
         await self.push_frame(frame, direction)
 
     def _update_sample_rate(self, frame: StartFrame):
-        self._sample_rate = self._init_sample_rate or frame.audio_out_sample_rate
+        # Record to the minimum sample rate to avoid possible downsampling
+        # artifacts.
+        min_sample_rate = min(frame.audio_in_sample_rate, frame.audio_out_sample_rate)
+        if frame.audio_in_sample_rate != frame.audio_out_sample_rate:
+            logger.debug(
+                f"{self} Input and output sample rates don't match, recording with smaller sample rate: {min_sample_rate} (this might get fixed in the future)"
+            )
+        self._sample_rate = self._init_sample_rate or min_sample_rate
         self._audio_buffer_size_1s = self._sample_rate * 2
 
     async def _process_recording(self, frame: Frame):