Sending audio faster than realtime.

This commit is contained in:
filipi87
2026-05-20 12:03:24 -03:00
parent aef6226a1c
commit 996aa461ac
2 changed files with 167 additions and 83 deletions

View File

@@ -1,6 +1,7 @@
import asyncio
import os
import signal
import time
from daily import (
AudioData,
@@ -15,6 +16,14 @@ from loguru import logger
load_dotenv(override=True)
# Pipecat sends audio at this true content rate but declares it as
# DECLARED_SAMPLE_RATE to write_frames(), which makes delivery faster than
# real-time. We receive at the declared rate (no resampling) and play back at
# the true rate so the avatar consumes audio at normal speed.
TRUE_SAMPLE_RATE = 24000
DECLARED_SAMPLE_RATE = 48000
SPEEDUP = DECLARED_SAMPLE_RATE // TRUE_SAMPLE_RATE
def completion_callback(future):
def _callback(*args):
@@ -37,19 +46,21 @@ class DailyProxyApp(EventHandler):
def __new__(cls, *args, **kwargs):
return super().__new__(cls)
def __init__(self, sample_rate: int):
def __init__(self):
super().__init__()
self._sample_rate = sample_rate
self._loop = asyncio.new_event_loop()
self._audio_queue: asyncio.Queue = asyncio.Queue()
# Raw PCM buffer — filled at DECLARED_SAMPLE_RATE speed, drained at TRUE_SAMPLE_RATE speed.
self._buffer = bytearray()
self._audio_task: asyncio.Task | None = None
self._receive_start_time: float | None = None
self._client: CallClient = CallClient(event_handler=self)
self._client.update_subscription_profiles(
{"base": {"camera": "unsubscribed", "microphone": "subscribed"}}
)
self._audio_source = CustomAudioSource(self._sample_rate, 1)
# Playback source declared at TRUE_SAMPLE_RATE — consumes audio at real-time speed.
self._audio_source = CustomAudioSource(TRUE_SAMPLE_RATE, 1)
self._audio_track = CustomAudioTrack(self._audio_source)
def on_joined(self, data, error):
@@ -113,7 +124,6 @@ class DailyProxyApp(EventHandler):
if self._audio_task:
self._audio_task.cancel()
try:
# Waits for it to finish
await self._audio_task
except asyncio.CancelledError:
pass
@@ -121,36 +131,60 @@ class DailyProxyApp(EventHandler):
async def capture_participant_audio(self, participant_id: str):
logger.info(f"Capturing participant audio: {participant_id}")
# Receiving from this custom track
# audio_source: str = "microphone"
audio_source: str = "stream"
media = {"media": {"customAudio": {audio_source: "subscribed"}}}
await self.update_subscriptions(participant_settings={participant_id: media})
# Must match the declared rate Pipecat used so WebRTC skips resampling —
# every original byte arrives intact.
self._client.set_audio_renderer(
participant_id,
self._audio_data_received,
audio_source=audio_source,
sample_rate=self._sample_rate,
sample_rate=DECLARED_SAMPLE_RATE,
callback_interval_ms=20,
)
logger.info(
f"Receiving at declared_rate={DECLARED_SAMPLE_RATE} Hz "
f"(true content: {TRUE_SAMPLE_RATE} Hz, ~{SPEEDUP}x faster than real-time)"
)
async def send_audio(self, audio: AudioData):
future = asyncio.get_running_loop().create_future()
self._audio_source.write_frames(audio.audio_frames, completion=completion_callback(future))
await future
async def _buffer_audio(self, audio_data: AudioData):
"""Append received bytes to the buffer and log the fill rate."""
new_bytes = audio_data.audio_frames
if self._receive_start_time is None:
self._receive_start_time = time.monotonic()
async def queue_audio(self, audio: AudioData):
await self._audio_queue.put(audio)
self._buffer.extend(new_bytes)
def _audio_data_received(self, participant_id: str, audio_data: AudioData, audio_source: str):
# logger.info(f"Received audio data for {participant_id}, audio_source: {audio_source}")
asyncio.run_coroutine_threadsafe(self.queue_audio(audio_data), self._loop)
asyncio.run_coroutine_threadsafe(self._buffer_audio(audio_data), self._loop)
async def _audio_task_handler(self):
"""Drain the buffer at TRUE_SAMPLE_RATE speed (real-time playback)."""
chunk_frames = int(TRUE_SAMPLE_RATE * 20 / 1000) # 20 ms chunks
chunk_bytes = chunk_frames * 2 # 16-bit mono
last_log_time = self._loop.time()
while True:
audio = await self._audio_queue.get()
await self.send_audio(audio)
if len(self._buffer) >= chunk_bytes:
chunk = bytes(self._buffer[:chunk_bytes])
del self._buffer[:chunk_bytes]
future = asyncio.get_running_loop().create_future()
self._audio_source.write_frames(chunk, completion=completion_callback(future))
await future
else:
await asyncio.sleep(0.001)
now = self._loop.time()
if now - last_log_time >= 1.0:
buffer_seconds = len(self._buffer) / (TRUE_SAMPLE_RATE * 2)
if buffer_seconds > 0:
logger.info(
f"Buffer status: {len(self._buffer)}B ({buffer_seconds:.3f}s buffered)"
)
last_log_time = now
#
# Daily (EventHandler)
@@ -160,7 +194,7 @@ class DailyProxyApp(EventHandler):
participant_name = participant["info"]["userName"]
logger.info(f"Participant {participant_name} joined")
if participant_name != "Pipecat":
# We are only subscribing for audios from Pipecat.
# We are only subscribing for audio from Pipecat.
return
asyncio.run_coroutine_threadsafe(
self.capture_participant_audio(participant_id=participant["id"]), self._loop
@@ -173,7 +207,7 @@ class DailyProxyApp(EventHandler):
def main():
Daily.init()
room_url = os.environ["TAVUS_SAMPLE_ROOM_URL"]
app = DailyProxyApp(sample_rate=24000)
app = DailyProxyApp()
app.run(room_url)