deepgeek/gradio-webrtc
2
0
Fork 0
mirror of https://github.com/HumanAIGC-Engineering/gradio-webrtc.git synced 2026-02-05 18:09:23 +08:00
Code Issues Packages Projects Releases Wiki Activity

Set ice candidates server (#285)

Browse Source 
* Add code

* Add code

* Code
This commit is contained in:
Freddy Boulton
2025-04-17 10:20:53 -04:00
committed by GitHub
parent 72fdc74e82
commit c9bca428af
7 changed files with 59 additions and 21 deletions
Download Patch File Download Diff File Expand all files Collapse all files

18
backend/fastrtc/webrtc.py
View File

@@ -2,7 +2,6 @@
from __future__ import annotations
import inspect
import logging
from collections.abc import Callable, Iterable, Sequence
from typing import (
@@ -15,8 +14,6 @@ from typing import (
cast,
)
import anyio
import anyio.to_thread
from gradio import wasm_utils
from gradio.components.base import Component, server
from gradio_client import handle_file
@@ -82,6 +79,7 @@ class WebRTC(Component, WebRTCConnectionMixin):
key: int | str | None = None,
mirror_webcam: bool = True,
rtc_configuration: dict[str, Any] | None | RTCConfigurationCallable = None,
server_rtc_configuration: dict[str, Any] | None = None,
track_constraints: dict[str, Any] | None = None,
time_limit: float | None = None,
allow_extra_tracks: bool = False,
@@ -115,6 +113,8 @@ class WebRTC(Component, WebRTCConnectionMixin):
key: if assigned, will be used to assume identity across a re-render. Components that have the same key across a re-render will have their value preserved.
mirror_webcam: if True webcam will be mirrored. Default is True.
rtc_configuration: WebRTC configuration options. See https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/RTCPeerConnection . If running the demo on a remote server, you will need to specify a rtc_configuration. See https://freddyaboulton.github.io/gradio-webrtc/deployment/
server_rtc_configuration: Optional dictionary for RTCPeerConnection configuration on the server side. Note
that setting iceServers to be an empty list will mean no ICE servers will be used in the server.
track_constraints: Media track constraints for WebRTC. For example, to set video height, width use {"width": {"exact": 800}, "height": {"exact": 600}, "aspectRatio": {"exact": 1.33333}}
time_limit: Maximum duration in seconds for recording.
allow_extra_tracks: Allow tracks not supported by the modality. For example, a peer connection with an audio track would be allowed even if modality is 'video', which normally throws a ``ValueError`` exception.
@@ -134,6 +134,9 @@ class WebRTC(Component, WebRTCConnectionMixin):
self.mirror_webcam = mirror_webcam
self.concurrency_limit = 1
self.rtc_configuration = rtc_configuration
self.server_rtc_configuration = self.convert_to_aiortc_format(
server_rtc_configuration
)
self.allow_extra_tracks = allow_extra_tracks
self.mode = mode
self.modality = modality
@@ -238,7 +241,6 @@ class WebRTC(Component, WebRTCConnectionMixin):
inputs = list(inputs)
async def handler(webrtc_id: str, *args):
print("webrtc_id", webrtc_id)
async for next_outputs in self.output_stream(webrtc_id):
yield fn(*args, *next_outputs.args) # type: ignore
@@ -366,13 +368,7 @@ class WebRTC(Component, WebRTCConnectionMixin):
@server
async def turn(self, _):
try:
if inspect.isfunction(self.rtc_configuration):
if inspect.iscoroutinefunction(self.rtc_configuration):
return await self.rtc_configuration()
else:
return await anyio.to_thread.run_sync(self.rtc_configuration)
else:
return self.rtc_configuration or {}
return await self.resolve_rtc_configuration()
except Exception as e:
return {"error": str(e)}