Fix audio buffer flush and silence handling
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@@ -229,9 +229,12 @@ class AudioBufferProcessor(FrameProcessor):
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# Save time of frame so we can compute silence.
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self._last_bot_frame_at = time.time()
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if self._buffer_size > 0 and len(self._user_audio_buffer) > self._buffer_size:
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if self._buffer_size > 0 and (
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len(self._user_audio_buffer) >= self._buffer_size
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or len(self._bot_audio_buffer) >= self._buffer_size
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):
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await self._call_on_audio_data_handler()
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self._reset_recording()
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self._clear_primary_audio_buffers()
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# Process turn recording with preprocessed data.
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if self._enable_turn_audio:
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@@ -272,9 +275,15 @@ class AudioBufferProcessor(FrameProcessor):
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async def _call_on_audio_data_handler(self):
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"""Call the audio data event handlers with buffered audio."""
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if not self.has_audio() or not self._recording:
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if not self._recording:
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return
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if len(self._user_audio_buffer) == 0 and len(self._bot_audio_buffer) == 0:
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return
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self._align_track_buffers()
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flush_time = time.time()
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# Call original handler with merged audio
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merged_audio = self.merge_audio_buffers()
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await self._call_event_handler(
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@@ -290,6 +299,9 @@ class AudioBufferProcessor(FrameProcessor):
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self._num_channels,
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)
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self._last_user_frame_at = flush_time
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self._last_bot_frame_at = flush_time
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def _buffer_has_audio(self, buffer: bytearray) -> bool:
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"""Check if a buffer contains audio data."""
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return buffer is not None and len(buffer) > 0
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@@ -307,6 +319,26 @@ class AudioBufferProcessor(FrameProcessor):
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self._user_turn_audio_buffer = bytearray()
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self._bot_turn_audio_buffer = bytearray()
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def _clear_primary_audio_buffers(self):
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"""Clear user and bot buffers while preserving turn buffers and timestamps."""
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self._user_audio_buffer = bytearray()
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self._bot_audio_buffer = bytearray()
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def _align_track_buffers(self):
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"""Pad the shorter track with silence so both tracks stay in sync."""
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user_len = len(self._user_audio_buffer)
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bot_len = len(self._bot_audio_buffer)
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if user_len == bot_len:
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return
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target_len = max(user_len, bot_len)
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if user_len < target_len:
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self._user_audio_buffer.extend(b"\x00" * (target_len - user_len))
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self._last_user_frame_at = max(self._last_user_frame_at, self._last_bot_frame_at)
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if bot_len < target_len:
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self._bot_audio_buffer.extend(b"\x00" * (target_len - bot_len))
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self._last_bot_frame_at = max(self._last_bot_frame_at, self._last_user_frame_at)
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async def _resample_input_audio(self, frame: InputAudioRawFrame) -> bytes:
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"""Resample audio frame to the target sample rate."""
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return await self._input_resampler.resample(
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