diff --git a/src/pipecat/audio/resamplers/base_audio_resampler.py b/src/pipecat/audio/resamplers/base_audio_resampler.py
index 0d7c3afef..6afbbcbfe 100644
--- a/src/pipecat/audio/resamplers/base_audio_resampler.py
+++ b/src/pipecat/audio/resamplers/base_audio_resampler.py
@@ -13,7 +13,7 @@ class BaseAudioResampler(ABC):
     """
 
     @abstractmethod
-    def resample(self, audio: bytes, in_rate: int, out_rate: int) -> bytes:
+    async def resample(self, audio: bytes, in_rate: int, out_rate: int) -> bytes:
         """
         Resamples the given audio data to a different sample rate.
 
diff --git a/src/pipecat/audio/resamplers/resampy_resampler.py b/src/pipecat/audio/resamplers/resampy_resampler.py
index be194b767..8c053fc3b 100644
--- a/src/pipecat/audio/resamplers/resampy_resampler.py
+++ b/src/pipecat/audio/resamplers/resampy_resampler.py
@@ -16,9 +16,10 @@ class ResampyResampler(BaseAudioResampler):
     def __init__(self, **kwargs):
         pass
 
-    def resample(self, audio: bytes, in_rate: int, out_rate: int) -> bytes:
+    async def resample(self, audio: bytes, in_rate: int, out_rate: int) -> bytes:
         if in_rate == out_rate:
             return audio
         audio_data = np.frombuffer(audio, dtype=np.int16)
         resampled_audio = resampy.resample(audio_data, in_rate, out_rate, filter="kaiser_fast")
-        return resampled_audio.astype(np.int16).tobytes()
+        result = resampled_audio.astype(np.int16).tobytes()
+        return result
diff --git a/src/pipecat/audio/resamplers/soxr_resampler.py b/src/pipecat/audio/resamplers/soxr_resampler.py
index eaa06ad4e..88edb84eb 100644
--- a/src/pipecat/audio/resamplers/soxr_resampler.py
+++ b/src/pipecat/audio/resamplers/soxr_resampler.py
@@ -16,9 +16,10 @@ class SOXRAudioResampler(BaseAudioResampler):
     def __init__(self, **kwargs):
         pass
 
-    def resample(self, audio: bytes, in_rate: int, out_rate: int) -> bytes:
+    async def resample(self, audio: bytes, in_rate: int, out_rate: int) -> bytes:
         if in_rate == out_rate:
             return audio
         audio_data = np.frombuffer(audio, dtype=np.int16)
         resampled_audio = soxr.resample(audio_data, in_rate, out_rate, quality="VHQ")
-        return resampled_audio.astype(np.int16).tobytes()
+        result = resampled_audio.astype(np.int16).tobytes()
+        return result
diff --git a/src/pipecat/audio/utils.py b/src/pipecat/audio/utils.py
index 9a9d442dd..0cb5d4f06 100644
--- a/src/pipecat/audio/utils.py
+++ b/src/pipecat/audio/utils.py
@@ -91,19 +91,21 @@ def exp_smoothing(value: float, prev_value: float, factor: float) -> float:
     return prev_value + factor * (value - prev_value)
 
 
-def ulaw_to_pcm(ulaw_bytes: bytes, in_rate: int, out_rate: int, resampler: BaseAudioResampler):
+async def ulaw_to_pcm(
+    ulaw_bytes: bytes, in_rate: int, out_rate: int, resampler: BaseAudioResampler
+):
     # Convert μ-law to PCM
     in_pcm_bytes = audioop.ulaw2lin(ulaw_bytes, 2)
 
     # Resample
-    out_pcm_bytes = resampler.resample(in_pcm_bytes, in_rate, out_rate)
+    out_pcm_bytes = await resampler.resample(in_pcm_bytes, in_rate, out_rate)
 
     return out_pcm_bytes
 
 
-def pcm_to_ulaw(pcm_bytes: bytes, in_rate: int, out_rate: int, resampler: BaseAudioResampler):
+async def pcm_to_ulaw(pcm_bytes: bytes, in_rate: int, out_rate: int, resampler: BaseAudioResampler):
     # Resample
-    in_pcm_bytes = resampler.resample(pcm_bytes, in_rate, out_rate)
+    in_pcm_bytes = await resampler.resample(pcm_bytes, in_rate, out_rate)
 
     # Convert PCM to μ-law
     ulaw_bytes = audioop.lin2ulaw(in_pcm_bytes, 2)
@@ -111,21 +113,21 @@ def pcm_to_ulaw(pcm_bytes: bytes, in_rate: int, out_rate: int, resampler: BaseAu
     return ulaw_bytes
 
 
-def alaw_to_pcm(
+async def alaw_to_pcm(
     alaw_bytes: bytes, in_rate: int, out_rate: int, resampler: BaseAudioResampler
 ) -> bytes:
     # Convert a-law to PCM
     in_pcm_bytes = audioop.alaw2lin(alaw_bytes, 2)
 
     # Resample
-    out_pcm_bytes = resampler.resample(in_pcm_bytes, in_rate, out_rate)
+    out_pcm_bytes = await resampler.resample(in_pcm_bytes, in_rate, out_rate)
 
     return out_pcm_bytes
 
 
-def pcm_to_alaw(pcm_bytes: bytes, in_rate: int, out_rate: int, resampler: BaseAudioResampler):
+async def pcm_to_alaw(pcm_bytes: bytes, in_rate: int, out_rate: int, resampler: BaseAudioResampler):
     # Resample
-    in_pcm_bytes = resampler.resample(pcm_bytes, in_rate, out_rate)
+    in_pcm_bytes = await resampler.resample(pcm_bytes, in_rate, out_rate)
 
     # Convert PCM to μ-law
     alaw_bytes = audioop.lin2alaw(in_pcm_bytes, 2)
diff --git a/src/pipecat/processors/audio/audio_buffer_processor.py b/src/pipecat/processors/audio/audio_buffer_processor.py
index 376f6bf74..f00da6eb8 100644
--- a/src/pipecat/processors/audio/audio_buffer_processor.py
+++ b/src/pipecat/processors/audio/audio_buffer_processor.py
@@ -77,7 +77,7 @@ class AudioBufferProcessor(FrameProcessor):
 
         # Include all audio from the user.
         if isinstance(frame, InputAudioRawFrame):
-            resampled = self._resample_audio(frame)
+            resampled = await self._resample_audio(frame)
             self._user_audio_buffer.extend(resampled)
             # Sync the bot's buffer to the user's buffer by adding silence if needed.
             if len(self._user_audio_buffer) > len(self._bot_audio_buffer):
@@ -86,7 +86,7 @@ class AudioBufferProcessor(FrameProcessor):
                 self._bot_audio_buffer.extend(silence)
         # If the bot is speaking, include all audio from the bot.
         elif isinstance(frame, OutputAudioRawFrame):
-            resampled = self._resample_audio(frame)
+            resampled = await self._resample_audio(frame)
             self._bot_audio_buffer.extend(resampled)
             # Sync the user's buffer to the bot's buffer by adding silence if needed.
             if len(self._bot_audio_buffer) > len(self._user_audio_buffer):
@@ -115,5 +115,5 @@ class AudioBufferProcessor(FrameProcessor):
     def _buffer_has_audio(self, buffer: bytearray) -> bool:
         return buffer is not None and len(buffer) > 0
 
-    def _resample_audio(self, frame: AudioRawFrame) -> bytes:
-        return self._resampler.resample(frame.audio, frame.sample_rate, self._sample_rate)
+    async def _resample_audio(self, frame: AudioRawFrame) -> bytes:
+        return await self._resampler.resample(frame.audio, frame.sample_rate, self._sample_rate)
diff --git a/src/pipecat/serializers/telnyx.py b/src/pipecat/serializers/telnyx.py
index 2e0246ea7..79c6f42ff 100644
--- a/src/pipecat/serializers/telnyx.py
+++ b/src/pipecat/serializers/telnyx.py
@@ -6,7 +6,6 @@
 
 import base64
 import json
-from typing import Optional
 
 from pydantic import BaseModel
 
@@ -58,11 +57,11 @@ class TelnyxFrameSerializer(FrameSerializer):
             data = frame.audio
 
             if self._params.inbound_encoding == "PCMU":
-                serialized_data = pcm_to_ulaw(
+                serialized_data = await pcm_to_ulaw(
                     data, frame.sample_rate, self._params.telnyx_sample_rate, self._resampler
                 )
             elif self._params.inbound_encoding == "PCMA":
-                serialized_data = pcm_to_alaw(
+                serialized_data = await pcm_to_alaw(
                     data, frame.sample_rate, self._params.telnyx_sample_rate, self._resampler
                 )
             else:
@@ -88,14 +87,14 @@ class TelnyxFrameSerializer(FrameSerializer):
             payload = base64.b64decode(payload_base64)
 
             if self._params.outbound_encoding == "PCMU":
-                deserialized_data = ulaw_to_pcm(
+                deserialized_data = await ulaw_to_pcm(
                     payload,
                     self._params.telnyx_sample_rate,
                     self._params.sample_rate,
                     self._resampler,
                 )
             elif self._params.outbound_encoding == "PCMA":
-                deserialized_data = alaw_to_pcm(
+                deserialized_data = await alaw_to_pcm(
                     payload,
                     self._params.telnyx_sample_rate,
                     self._params.sample_rate,
diff --git a/src/pipecat/serializers/twilio.py b/src/pipecat/serializers/twilio.py
index c6d4e1e13..7862fa0db 100644
--- a/src/pipecat/serializers/twilio.py
+++ b/src/pipecat/serializers/twilio.py
@@ -45,7 +45,7 @@ class TwilioFrameSerializer(FrameSerializer):
         elif isinstance(frame, AudioRawFrame):
             data = frame.audio
 
-            serialized_data = pcm_to_ulaw(
+            serialized_data = await pcm_to_ulaw(
                 data, frame.sample_rate, self._params.twilio_sample_rate, self._resampler
             )
             payload = base64.b64encode(serialized_data).decode("utf-8")
@@ -66,7 +66,7 @@ class TwilioFrameSerializer(FrameSerializer):
             payload_base64 = message["media"]["payload"]
             payload = base64.b64decode(payload_base64)
 
-            deserialized_data = ulaw_to_pcm(
+            deserialized_data = await ulaw_to_pcm(
                 payload, self._params.twilio_sample_rate, self._params.sample_rate, self._resampler
             )
             audio_frame = InputAudioRawFrame(
diff --git a/src/pipecat/services/aws.py b/src/pipecat/services/aws.py
index eb03bbc59..4444f07b7 100644
--- a/src/pipecat/services/aws.py
+++ b/src/pipecat/services/aws.py
@@ -195,10 +195,7 @@ class PollyTTSService(TTSService):
             response = self._polly_client.synthesize_speech(**args)
             if "AudioStream" in response:
                 audio_data = response["AudioStream"].read()
-                resampled = self._resampler.resample(
-                    audio_data, 16000, self._settings["sample_rate"]
-                )
-                return resampled
+                return audio_data
             return None
 
         logger.debug(f"Generating TTS: [{text}]")
@@ -229,6 +226,10 @@ class PollyTTSService(TTSService):
                 yield None
                 return
 
+            audio_data = await self._resampler.resample(
+                audio_data, 16000, self._settings["sample_rate"]
+            )
+
             await self.start_tts_usage_metrics(text)
 
             yield TTSStartedFrame()
diff --git a/src/pipecat/services/tavus.py b/src/pipecat/services/tavus.py
index e6693ecc7..b0ca699cb 100644
--- a/src/pipecat/services/tavus.py
+++ b/src/pipecat/services/tavus.py
@@ -94,7 +94,7 @@ class TavusVideoService(AIService):
     async def _encode_audio_and_send(self, audio: bytes, in_rate: int, done: bool) -> None:
         """Encodes audio to base64 and sends it to Tavus"""
         if not done:
-            audio = self._resampler.resample(audio, in_rate, 16000)
+            audio = await self._resampler.resample(audio, in_rate, 16000)
         audio_base64 = base64.b64encode(audio).decode("utf-8")
         logger.trace(f"{self}: sending {len(audio)} bytes")
         await self._send_audio_message(audio_base64, done=done)
diff --git a/src/pipecat/services/xtts.py b/src/pipecat/services/xtts.py
index b26660175..a10247acd 100644
--- a/src/pipecat/services/xtts.py
+++ b/src/pipecat/services/xtts.py
@@ -163,7 +163,7 @@ class XTTSService(TTSService):
                         buffer = buffer[48000:]
 
                         # XTTS uses 24000 so we need to resample to our desired rate.
-                        resampled_audio = self._resampler.resample(
+                        resampled_audio = await self._resampler.resample(
                             bytes(process_data), 24000, self._sample_rate
                         )
                         # Create the frame with the resampled audio
@@ -172,7 +172,9 @@ class XTTSService(TTSService):
 
             # Process any remaining data in the buffer.
             if len(buffer) > 0:
-                resampled_audio = self._resampler.resample(bytes(buffer), 24000, self._sample_rate)
+                resampled_audio = await self._resampler.resample(
+                    bytes(buffer), 24000, self._sample_rate
+                )
                 frame = TTSAudioRawFrame(resampled_audio, self._sample_rate, 1)
                 yield frame