TTSService: resample _stream_audio_frames_from_iterator() input audio if needed

src/pipecat/services/tts_service.py

@@ -24,6 +24,7 @@ from typing import (
 
 from loguru import logger
 
+from pipecat.audio.utils import create_stream_resampler
 from pipecat.frames.frames import (
     AggregatedTextFrame,
     AggregationType,
@@ -202,6 +203,8 @@ class TTSService(AIService):
                 )
             self._text_filters = [text_filter]
 
+        self._resampler = create_stream_resampler()
+
         self._stop_frame_task: Optional[asyncio.Task] = None
         self._stop_frame_queue: asyncio.Queue = asyncio.Queue()
 
@@ -505,12 +508,40 @@ class TTSService(AIService):
             await self._stop_frame_queue.put(frame)
 
     async def _stream_audio_frames_from_iterator(
-        self, iterator: AsyncIterator[bytes], *, strip_wav_header: bool
+        self,
+        iterator: AsyncIterator[bytes],
+        *,
+        strip_wav_header: bool = False,
+        in_sample_rate: Optional[int] = None,
     ) -> AsyncGenerator[Frame, None]:
+        """Stream audio frames from an async byte iterator with optional resampling.
+
+        For WAV data, use `strip_wav_header=True` to strip the header and
+        auto-detect the source sample rate. For raw PCM data, pass
+        `in_sample_rate` directly. Audio is resampled to `self.sample_rate` when
+        the source rate differs.
+
+        Args:
+            iterator: Async iterator yielding audio bytes.
+            strip_wav_header: Strip WAV header and parse source sample rate from it.
+            in_sample_rate: Source sample rate for raw PCM data. Overrides
+                WAV-detected rate if both are provided.
+
+        """
         buffer = bytearray()
+        source_sample_rate = in_sample_rate
         need_to_strip_wav_header = strip_wav_header
+
+        async def maybe_resample(audio: bytes) -> bytes:
+            if source_sample_rate and source_sample_rate != self.sample_rate:
+                return await self._resampler.resample(audio, source_sample_rate, self.sample_rate)
+            return audio
+
         async for chunk in iterator:
             if need_to_strip_wav_header and chunk.startswith(b"RIFF"):
+                # Parse sample rate from WAV header (bytes 24-28, little-endian uint32).
+                if len(chunk) >= 44 and source_sample_rate is None:
+                    source_sample_rate = int.from_bytes(chunk[24:28], "little")
                 chunk = chunk[44:]
                 need_to_strip_wav_header = False
 
@@ -520,19 +551,18 @@ class TTSService(AIService):
             # Round to nearest even number.
             aligned_length = len(buffer) & ~1  # 111111111...11110
             if aligned_length > 0:
-                aligned_chunk = buffer[:aligned_length]
+                aligned_chunk = await maybe_resample(bytes(buffer[:aligned_length]))
                 buffer = buffer[aligned_length:]  # keep any leftover byte
 
                 if len(aligned_chunk) > 0:
-                    frame = TTSAudioRawFrame(bytes(aligned_chunk), self.sample_rate, 1)
-                    yield frame
+                    yield TTSAudioRawFrame(aligned_chunk, self.sample_rate, 1)
 
         if len(buffer) > 0:
             # Make sure we don't need an extra padding byte.
             if len(buffer) % 2 == 1:
                 buffer.extend(b"\x00")
-            frame = TTSAudioRawFrame(bytes(buffer), self.sample_rate, 1)
-            yield frame
+            audio = await maybe_resample(bytes(buffer))
+            yield TTSAudioRawFrame(audio, self.sample_rate, 1)
 
     async def _handle_interruption(self, frame: InterruptionFrame, direction: FrameDirection):
         self._processing_text = False