Files
gradio-webrtc/backend/gradio_webrtc/utils.py
freddyaboulton e87c4d49e8 final touches
2024-10-23 15:52:48 -07:00

83 lines
2.5 KiB
Python

import asyncio
import fractions
import logging
from typing import Callable
import av
logger = logging.getLogger(__name__)
AUDIO_PTIME = 0.020
async def player_worker_decode(
next_frame: Callable,
queue: asyncio.Queue,
thread_quit: asyncio.Event,
quit_on_none: bool = False,
sample_rate: int = 48000,
frame_size: int = int(48000 * AUDIO_PTIME),
):
audio_samples = 0
audio_time_base = fractions.Fraction(1, sample_rate)
audio_resampler = av.AudioResampler( # type: ignore
format="s16",
layout="stereo",
rate=sample_rate,
frame_size=frame_size,
)
while not thread_quit.is_set():
try:
async with asyncio.timeout(5):
# Get next frame
frame = await next_frame()
if frame is None:
if quit_on_none:
await queue.put(None)
break
continue
if len(frame) == 2:
sample_rate, audio_array = frame
layout = "mono"
elif len(frame) == 3:
sample_rate, audio_array, layout = frame
logger.debug(
"received array with shape %s sample rate %s layout %s",
audio_array.shape,
sample_rate,
layout,
)
format = "s16" if audio_array.dtype == "int16" else "fltp"
# Convert to audio frame and resample
# This runs in the same timeout context
frame = av.AudioFrame.from_ndarray(
audio_array, format=format, layout=layout
)
frame.sample_rate = sample_rate
for processed_frame in audio_resampler.resample(frame):
processed_frame.pts = audio_samples
processed_frame.time_base = audio_time_base
audio_samples += processed_frame.samples
await queue.put(processed_frame)
logger.debug("Queue size utils.py: %s", queue.qsize())
except TimeoutError:
logger.warning(
"Timeout in frame processing cycle after %s seconds - resetting", 5
)
continue
except Exception as e:
import traceback
exec = traceback.format_exc()
logger.debug("traceback %s", exec)
logger.error("Error processing frame: %s", str(e))
continue