tts: make frame pausing/resuming optional

src/pipecat/services/ai_services.py

@@ -76,13 +76,13 @@ class AIService(FrameProcessor):
         )
 
         for key, value in settings.items():
-            print("Update request for:", key, value)
+            logger.debug("Update request for:", key, value)
 
             if key in self._settings:
                 logger.info(f"Updating LLM setting {key} to: [{value}]")
                 self._settings[key] = value
             elif key in SessionProperties.model_fields:
-                print("Attempting to update", key, value)
+                logger.debug("Attempting to update", key, value)
 
                 try:
                     from pipecat.services.openai_realtime_beta.events import (
@@ -213,6 +213,8 @@ class TTSService(AIService):
         push_silence_after_stop: bool = False,
         # if push_silence_after_stop is True, send this amount of audio silence
         silence_time_s: float = 2.0,
+        # if True, we will pause processing frames while we are receiving audio
+        pause_frame_processing: bool = False,
         # TTS output sample rate
         sample_rate: Optional[int] = None,
         text_filter: Optional[BaseTextFilter] = None,
@@ -225,6 +227,7 @@ class TTSService(AIService):
         self._stop_frame_timeout_s: float = stop_frame_timeout_s
         self._push_silence_after_stop: bool = push_silence_after_stop
         self._silence_time_s: float = silence_time_s
+        self._pause_frame_processing: bool = pause_frame_processing
         self._init_sample_rate = sample_rate
         self._sample_rate = 0
         self._voice_id: str = ""
@@ -314,8 +317,7 @@ class TTSService(AIService):
             # We pause processing incoming frames if the LLM response included
             # text (it might be that it's only a function calling response). We
             # pause to avoid audio overlapping.
-            if self._processing_text:
-                await self.pause_processing_frames()
+            await self._maybe_pause_frame_processing()
 
             sentence = self._current_sentence
             self._current_sentence = ""
@@ -327,16 +329,16 @@ class TTSService(AIService):
             else:
                 await self.push_frame(frame, direction)
         elif isinstance(frame, TTSSpeakFrame):
+            await self._push_tts_frames(frame.text)
             # We pause processing incoming frames because we are sending data to
             # the TTS. We pause to avoid audio overlapping.
-            await self.pause_processing_frames()
-            await self._push_tts_frames(frame.text)
+            await self._maybe_pause_frame_processing()
             await self.flush_audio()
             self._processing_text = False
         elif isinstance(frame, TTSUpdateSettingsFrame):
             await self._update_settings(frame.settings)
         elif isinstance(frame, BotStoppedSpeakingFrame):
-            await self.resume_processing_frames()
+            await self._maybe_resume_frame_processing()
         else:
             await self.push_frame(frame, direction)
 
@@ -367,6 +369,14 @@ class TTSService(AIService):
             self._text_filter.handle_interruption()
         await self.push_frame(frame, direction)
 
+    async def _maybe_pause_frame_processing(self):
+        if self._processing_text and self._pause_frame_processing:
+            await self.pause_processing_frames()
+
+    async def _maybe_resume_frame_processing(self):
+        if self._pause_frame_processing:
+            await self.resume_processing_frames()
+
     async def _process_text_frame(self, frame: TextFrame):
         text: Optional[str] = None
         if not self._aggregate_sentences:

src/pipecat/services/cartesia.py

@@ -109,6 +109,7 @@ class CartesiaTTSService(AudioContextWordTTSService, WebsocketService):
             self,
             aggregate_sentences=True,
             push_text_frames=False,
+            pause_frame_processing=True,
             sample_rate=sample_rate,
             **kwargs,
         )

src/pipecat/services/elevenlabs.py

@@ -192,6 +192,7 @@ class ElevenLabsTTSService(WordTTSService, WebsocketService):
             push_text_frames=False,
             push_stop_frames=True,
             stop_frame_timeout_s=2.0,
+            pause_frame_processing=True,
             sample_rate=sample_rate,
             **kwargs,
         )

src/pipecat/services/fish.py

@@ -60,7 +60,7 @@ class FishAudioTTSService(TTSService, WebsocketService):
         params: InputParams = InputParams(),
         **kwargs,
     ):
-        super().__init__(sample_rate=sample_rate, **kwargs)
+        super().__init__(pause_frame_processing=True, sample_rate=sample_rate, **kwargs)
 
         self._api_key = api_key
         self._base_url = "wss://api.fish.audio/v1/tts/live"

src/pipecat/services/lmnt.py

@@ -73,6 +73,7 @@ class LmntTTSService(TTSService, WebsocketService):
         TTSService.__init__(
             self,
             push_stop_frames=True,
+            pause_frame_processing=True,
             sample_rate=sample_rate,
             **kwargs,
         )

src/pipecat/services/playht.py

@@ -120,6 +120,7 @@ class PlayHTTTSService(TTSService, WebsocketService):
     ):
         TTSService.__init__(
             self,
+            pause_frame_processing=True,
             sample_rate=sample_rate,
             **kwargs,
         )

src/pipecat/services/rime.py

@@ -101,6 +101,7 @@ class RimeTTSService(AudioContextWordTTSService, WebsocketService):
             push_text_frames=False,
             push_stop_frames=True,
             stop_frame_timeout_s=2.0,
+            pause_frame_processing=True,
             sample_rate=sample_rate,
             **kwargs,
         )