Fixing the issue when using the TavusVideoService with some TTS services.

This commit is contained in:
Filipi Fuchter
2025-05-26 22:55:20 -03:00
parent 5a58357429
commit 2b3d2cb342

View File

@@ -170,15 +170,11 @@ class TavusVideoService(AIService):
await self._handle_interruptions()
await self.push_frame(frame, direction)
elif isinstance(frame, TTSStartedFrame):
await self.start_processing_metrics()
await self.start_ttfb_metrics()
self._current_idx_str = str(frame.id)
await self._queue.put(frame)
elif isinstance(frame, TTSAudioRawFrame):
await self._queue_audio(frame.audio, frame.sample_rate, done=False)
await self._queue.put(frame)
elif isinstance(frame, TTSStoppedFrame):
await self._queue_audio(b"\x00\x00", self._client.in_sample_rate, done=True)
await self.stop_ttfb_metrics()
await self.stop_processing_metrics()
await self._queue.put(frame)
else:
await self.push_frame(frame, direction)
@@ -191,9 +187,6 @@ class TavusVideoService(AIService):
await self._client.stop()
self._other_participant_has_joined = False
async def _queue_audio(self, audio: bytes, in_rate: int, done: bool):
await self._queue.put((audio, in_rate, done))
async def _create_send_task(self):
if not self._send_task:
self._queue = asyncio.Queue()
@@ -224,35 +217,35 @@ class TavusVideoService(AIService):
MAX_CHUNK_SIZE = int((sample_rate * 2) / 20)
audio_buffer = bytearray()
samples_sent = 0
start_time = time.time()
current_idx_str = None
silence = b"\x00\x00"
while True:
(audio, in_rate, done) = await self._queue.get()
if done:
frame = await self._queue.get()
if isinstance(frame, TTSStartedFrame):
if current_idx_str is not None:
continue
current_idx_str = str(frame.id)
elif isinstance(frame, TTSStoppedFrame):
# Send any remaining audio.
if len(audio_buffer) > 0:
await self._client.encode_audio_and_send(
bytes(audio_buffer), done, self._current_idx_str
bytes(audio_buffer), True, current_idx_str
)
await self._client.encode_audio_and_send(audio, done, self._current_idx_str)
await self._client.encode_audio_and_send(silence, True, current_idx_str)
audio_buffer.clear()
else:
audio = await self._resampler.resample(audio, in_rate, sample_rate)
current_idx_str = None
elif isinstance(frame, TTSAudioRawFrame):
if current_idx_str is None:
continue
audio = await self._resampler.resample(frame.audio, frame.sample_rate, sample_rate)
audio_buffer.extend(audio)
while len(audio_buffer) >= MAX_CHUNK_SIZE:
chunk = audio_buffer[:MAX_CHUNK_SIZE]
audio_buffer = audio_buffer[MAX_CHUNK_SIZE:]
# Compute wait time for synchronization
wait = start_time + (samples_sent / sample_rate) - time.time()
if wait > 0:
await asyncio.sleep(wait)
wait = 1 / 20 # 50ms
await asyncio.sleep(wait)
await self._client.encode_audio_and_send(
bytes(chunk), done, self._current_idx_str
)
# Update timestamp based on number of samples sent
samples_sent += len(chunk) // 2 # 2 bytes per sample (16-bit)
await self._client.encode_audio_and_send(bytes(chunk), False, current_idx_str)