Fix audio buffer flush and silence handling

This commit is contained in:
Jin Kim
2025-09-18 19:40:45 +09:00
parent 9e7260393a
commit d0b573e44f

View File

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