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Merge pull request #6 from freddyaboulton/use-logging-instead

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Fix send-receive video; Use logging instead of prints
This commit is contained in:
Freddy Boulton
2024-10-11 13:04:53 -07:00
committed by GitHub
parent d681dbfd7e 32c1be3f5b
commit 1533f98ed8
12 changed files with 213 additions and 241 deletions
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13
backend/gradio_webrtc/utils.py
View File

@@ -1,10 +1,15 @@
import time
import fractions
import av
import asyncio import asyncio
import fractions
import logging
import threading import threading
import time
from typing import Callable from typing import Callable
import av
logger = logging.getLogger(__name__)
AUDIO_PTIME = 0.020 AUDIO_PTIME = 0.020
@@ -39,7 +44,7 @@ def player_worker_decode(
frame = next(generator) frame = next(generator)
except Exception as exc: except Exception as exc:
if isinstance(exc, StopIteration): if isinstance(exc, StopIteration):
print("Not iterating") logger.debug("Stopping audio stream")
asyncio.run_coroutine_threadsafe(queue.put(None), loop) asyncio.run_coroutine_threadsafe(queue.put(None), loop)
thread_quit.set() thread_quit.set()
break break

137
backend/gradio_webrtc/webrtc.py
View File

@@ -1,31 +1,33 @@
"""gr.Video() component.""" """gr.WebRTC() component."""
from __future__ import annotations from __future__ import annotations
from abc import ABC, abstractmethod
import asyncio import asyncio
from collections.abc import Callable, Sequence import logging
from typing import TYPE_CHECKING, Any, Literal, cast, Generator
import fractions
import threading import threading
import time import time
from gradio_webrtc.utils import player_worker_decode import traceback
from collections.abc import Callable
from typing import TYPE_CHECKING, Any, Generator, Literal, Sequence, cast
from aiortc import RTCPeerConnection, RTCSessionDescription
from aiortc.contrib.media import MediaRelay
from aiortc import VideoStreamTrack, AudioStreamTrack
from aiortc.mediastreams import MediaStreamError
from aiortc.contrib.media import AudioFrame, VideoFrame # type: ignore
from gradio_client import handle_file
import numpy as np import numpy as np
from aiortc import (
AudioStreamTrack,
RTCPeerConnection,
RTCSessionDescription,
VideoStreamTrack,
)
from aiortc.contrib.media import MediaRelay, VideoFrame # type: ignore
from aiortc.mediastreams import MediaStreamError
from gradio import wasm_utils from gradio import wasm_utils
from gradio.components.base import Component, server from gradio.components.base import Component, server
from gradio_client import handle_file
from gradio_webrtc.utils import player_worker_decode
if TYPE_CHECKING: if TYPE_CHECKING:
from gradio.components import Timer
from gradio.blocks import Block from gradio.blocks import Block
from gradio.components import Timer
from gradio.events import Dependency from gradio.events import Dependency
@@ -33,6 +35,9 @@ if wasm_utils.IS_WASM:
raise ValueError("Not supported in gradio-lite!") raise ValueError("Not supported in gradio-lite!")
logger = logging.getLogger(__name__)
class VideoCallback(VideoStreamTrack): class VideoCallback(VideoStreamTrack):
""" """
This works for streaming input and output This works for streaming input and output
@@ -90,10 +95,9 @@ class VideoCallback(VideoStreamTrack):
return new_frame return new_frame
except Exception as e: except Exception as e:
print(e) logger.debug(e)
import traceback exec = traceback.format_exc()
logger.debug(exec)
traceback.print_exc()
class ServerToClientVideo(VideoStreamTrack): class ServerToClientVideo(VideoStreamTrack):
@@ -150,10 +154,9 @@ class ServerToClientVideo(VideoStreamTrack):
next_frame.time_base = time_base next_frame.time_base = time_base
return next_frame return next_frame
except Exception as e: except Exception as e:
print(e) logger.debug(e)
import traceback exec = traceback.format_exc()
logger.debug(exec)
traceback.print_exc()
class ServerToClientAudio(AudioStreamTrack): class ServerToClientAudio(AudioStreamTrack):
@@ -173,26 +176,6 @@ class ServerToClientAudio(AudioStreamTrack):
self._start: float | None = None self._start: float | None = None
super().__init__() super().__init__()
def array_to_frame(self, array: tuple[int, np.ndarray]) -> AudioFrame:
frame = AudioFrame.from_ndarray(array[1], format="s16", layout="mono")
frame.sample_rate = array[0]
frame.time_base = fractions.Fraction(1, array[0])
self.current_timestamp += array[1].shape[1]
frame.pts = self.current_timestamp
return frame
async def empty_frame(self) -> AudioFrame:
sample_rate = 22050
samples = 100
frame = AudioFrame(format="s16", layout="mono", samples=samples)
for p in frame.planes:
p.update(bytes(p.buffer_size))
frame.sample_rate = sample_rate
frame.time_base = fractions.Fraction(1, sample_rate)
self.current_timestamp += samples
frame.pts = self.current_timestamp
return frame
def start(self): def start(self):
if self.__thread is None: if self.__thread is None:
self.__thread = threading.Thread( self.__thread = threading.Thread(
@@ -232,10 +215,9 @@ class ServerToClientAudio(AudioStreamTrack):
return data return data
except Exception as e: except Exception as e:
print(e) logger.debug(e)
import traceback exec = traceback.format_exc()
logger.debug(exec)
traceback.print_exc()
def stop(self): def stop(self):
self.thread_quit.set() self.thread_quit.set()
@@ -244,39 +226,6 @@ class ServerToClientAudio(AudioStreamTrack):
self.__thread = None self.__thread = None
super().stop() super().stop()
# next_frame = await super().recv()
# print("next frame", next_frame)
# return next_frame
# try:
# if self.latest_args == "not_set":
# frame = await self.empty_frame()
# # await self.modify_frame(frame)
# await asyncio.sleep(100 / 22050)
# print("next_frame not set", frame)
# return frame
# if self.generator is None:
# self.generator = cast(
# Generator[Any, None, Any], self.event_handler(*self.latest_args)
# )
# try:
# next_array = next(self.generator)
# print("iteration")
# except StopIteration:
# print("exception")
# self.stop() # type: ignore
# return
# next_frame = self.array_to_frame(next_array)
# # await self.modify_frame(next_frame)
# print("next frame", next_frame)
# return next_frame
# except Exception as e:
# print(e)
# import traceback
# traceback.print_exc()
class WebRTC(Component): class WebRTC(Component):
""" """
@@ -485,7 +434,8 @@ class WebRTC(Component):
@server @server
async def offer(self, body): async def offer(self, body):
print("starting") logger.debug("Starting to handle offer")
logger.debug("Offer body", body)
if len(self.connections) >= cast(int, self.concurrency_limit): if len(self.connections) >= cast(int, self.concurrency_limit):
return {"status": "failed"} return {"status": "failed"}
@@ -496,7 +446,7 @@ class WebRTC(Component):
@pc.on("iceconnectionstatechange") @pc.on("iceconnectionstatechange")
async def on_iceconnectionstatechange(): async def on_iceconnectionstatechange():
print(pc.iceConnectionState) logger.debug("ICE connection state change", pc.iceConnectionState)
if pc.iceConnectionState == "failed": if pc.iceConnectionState == "failed":
await pc.close() await pc.close()
self.connections.pop(body["webrtc_id"], None) self.connections.pop(body["webrtc_id"], None)
@@ -519,32 +469,27 @@ class WebRTC(Component):
event_handler=cast(Callable, self.event_handler), event_handler=cast(Callable, self.event_handler),
) )
self.connections[body["webrtc_id"]] = cb self.connections[body["webrtc_id"]] = cb
logger.debug("Adding track to peer connection", cb)
pc.addTrack(cb) pc.addTrack(cb)
if self.mode == "receive" and self.modality == "video": if self.mode == "receive":
cb = ServerToClientVideo(cast(Callable, self.event_handler)) if self.modality == "video":
pc.addTrack(cb) cb = ServerToClientVideo(cast(Callable, self.event_handler))
self.connections[body["webrtc_id"]] = cb elif self.modality == "audio":
cb.on("ended", lambda: self.connections.pop(body["webrtc_id"], None)) cb = ServerToClientAudio(cast(Callable, self.event_handler))
if self.mode == "receive" and self.modality == "audio":
print("adding") logger.debug("Adding track to peer connection", cb)
cb = ServerToClientAudio(cast(Callable, self.event_handler))
print("cb.recv", cb.recv)
# from aiortc.contrib.media import MediaPlayer
# player = MediaPlayer("/Users/freddy/sources/gradio/demo/audio_debugger/cantina.wav")
# pc.addTrack(player.audio)
pc.addTrack(cb) pc.addTrack(cb)
self.connections[body["webrtc_id"]] = cb self.connections[body["webrtc_id"]] = cb
cb.on("ended", lambda: self.connections.pop(body["webrtc_id"], None)) cb.on("ended", lambda: self.connections.pop(body["webrtc_id"], None))
print("here")
# handle offer # handle offer
await pc.setRemoteDescription(offer) await pc.setRemoteDescription(offer)
# send answer # send answer
answer = await pc.createAnswer() answer = await pc.createAnswer()
await pc.setLocalDescription(answer) # type: ignore await pc.setLocalDescription(answer) # type: ignore
print("done") logger.debug("done handling offer about to return")
return { return {
"sdp": pc.localDescription.sdp, "sdp": pc.localDescription.sdp,

4
demo/audio_out.py
View File

@@ -22,18 +22,20 @@ if account_sid and auth_token:
else: else:
rtc_configuration = None rtc_configuration = None
import time
def generation(num_steps): def generation(num_steps):
for _ in range(num_steps): for _ in range(num_steps):
segment = AudioSegment.from_file("/Users/freddy/sources/gradio/demo/audio_debugger/cantina.wav") segment = AudioSegment.from_file("/Users/freddy/sources/gradio/demo/audio_debugger/cantina.wav")
yield (segment.frame_rate, np.array(segment.get_array_of_samples()).reshape(1, -1)) yield (segment.frame_rate, np.array(segment.get_array_of_samples()).reshape(1, -1))
time.sleep(3.5)
css = """.my-group {max-width: 600px !important; max-height: 600 !important;} css = """.my-group {max-width: 600px !important; max-height: 600 !important;}
.my-column {display: flex !important; justify-content: center !important; align-items: center !important};""" .my-column {display: flex !important; justify-content: center !important; align-items: center !important};"""
with gr.Blocks(css=css) as demo: with gr.Blocks() as demo:
gr.HTML( gr.HTML(
""" """
<h1 style='text-align: center'> <h1 style='text-align: center'>

5
demo/space.py
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File diff suppressed because one or more lines are too long

2
demo/video_out.py
View File

@@ -48,7 +48,7 @@ with gr.Blocks() as demo:
input_video = gr.Video(sources="upload") input_video = gr.Video(sources="upload")
with gr.Column(): with gr.Column():
output_video = WebRTC(label="Video Stream", rtc_configuration=rtc_configuration, output_video = WebRTC(label="Video Stream", rtc_configuration=rtc_configuration,
mode="receive", modality="video") mode="receive", modality="video")
output_video.stream( output_video.stream(
fn=generation, inputs=[input_video], outputs=[output_video], fn=generation, inputs=[input_video], outputs=[output_video],
trigger=input_video.upload trigger=input_video.upload

98
frontend/Index.svelte
View File

@@ -35,10 +35,73 @@
let dragging = false; let dragging = false;
$: console.log("value", value); $: console.log("value", value);
</script> </script>
<Block {#if mode == "receive" && modality === "video"}
<Block
{visible}
variant={"solid"}
border_mode={dragging ? "focus" : "base"}
padding={false}
{elem_id}
{elem_classes}
{height}
{width}
{container}
{scale}
{min_width}
allow_overflow={false}
>
<StatusTracker
autoscroll={gradio.autoscroll}
i18n={gradio.i18n}
{...loading_status}
on:clear_status={() => gradio.dispatch("clear_status", loading_status)}
/>
<StaticVideo
bind:value={value}
{label}
{show_label}
{server}
{rtc_configuration}
on:tick={() => gradio.dispatch("tick")}
on:error={({ detail }) => gradio.dispatch("error", detail)}
/>
</Block>
{:else if mode == "receive" && modality === "audio"}
<Block
variant={"solid"}
border_mode={dragging ? "focus" : "base"}
padding={false}
allow_overflow={false}
{elem_id}
{elem_classes}
{visible}
{container}
{scale}
{min_width}
>
<StatusTracker
autoscroll={gradio.autoscroll}
i18n={gradio.i18n}
{...loading_status}
on:clear_status={() => gradio.dispatch("clear_status", loading_status)}
/>
<StaticAudio
bind:value={value}
{label}
{show_label}
{server}
{rtc_configuration}
i18n={gradio.i18n}
on:tick={() => gradio.dispatch("tick")}
on:error={({ detail }) => gradio.dispatch("error", detail)}
/>
</Block>
{:else if mode === "send-receive" && modality === "video"}
<Block
{visible} {visible}
variant={"solid"} variant={"solid"}
border_mode={dragging ? "focus" : "base"} border_mode={dragging ? "focus" : "base"}
@@ -51,36 +114,13 @@
{scale} {scale}
{min_width} {min_width}
allow_overflow={false} allow_overflow={false}
> >
<StatusTracker <StatusTracker
autoscroll={gradio.autoscroll} autoscroll={gradio.autoscroll}
i18n={gradio.i18n} i18n={gradio.i18n}
{...loading_status} {...loading_status}
on:clear_status={() => gradio.dispatch("clear_status", loading_status)} on:clear_status={() => gradio.dispatch("clear_status", loading_status)}
/> />
{#if mode === "receive" && modality === "video"}
<StaticVideo
bind:value={value}
{label}
{show_label}
{server}
{rtc_configuration}
on:tick={() => gradio.dispatch("tick")}
on:error={({ detail }) => gradio.dispatch("error", detail)}
/>
{:else if mode == "receive" && modality === "audio"}
<StaticAudio
bind:value={value}
{label}
{show_label}
{server}
{rtc_configuration}
i18n={gradio.i18n}
on:tick={() => gradio.dispatch("tick")}
on:error={({ detail }) => gradio.dispatch("error", detail)}
/>
{:else if mode === "send-receive" && modality === "video"}
<Video <Video
bind:value={value} bind:value={value}
{label} {label}
@@ -105,5 +145,5 @@
> >
<UploadText i18n={gradio.i18n} type="video" /> <UploadText i18n={gradio.i18n} type="video" />
</Video> </Video>
{/if} </Block>
</Block> {/if}

3
frontend/shared/InteractiveVideo.svelte
View File

@@ -38,7 +38,8 @@
let dragging = false; let dragging = false;
$: dispatch("drag", dragging); $: dispatch("drag", dragging);
$: console.log("interactive value", value); $: console.log("value", value)
</script> </script>
<BlockLabel {show_label} Icon={Video} label={label || "Video"} /> <BlockLabel {show_label} Icon={Video} label={label || "Video"} />

42
frontend/shared/StaticAudio.svelte
View File

@@ -49,34 +49,27 @@
$: if( value === "start_webrtc_stream") { $: if( value === "start_webrtc_stream") {
stream_state = "connecting"; stream_state = "connecting";
value = _webrtc_id; value = _webrtc_id;
const fallback_config = { pc = new RTCPeerConnection(rtc_configuration);
iceServers: [ pc.addEventListener("connectionstatechange",
{ async (event) => {
urls: 'stun:stun.l.google.com:19302' switch(pc.connectionState) {
} case "connected":
] console.info("connected");
}; stream_state = "open";
pc = new RTCPeerConnection(rtc_configuration); break;
pc.addEventListener("connectionstatechange", case "disconnected":
async (event) => { console.info("closed");
switch(pc.connectionState) { stop(pc);
case "connected": break;
console.log("connected"); default:
stream_state = "open"; break;
break;
case "disconnected":
console.log("closed");
stop(pc);
break;
default:
break;
}
} }
) }
)
start(null, pc, audio_player, server.offer, _webrtc_id, "audio").then((connection) => { start(null, pc, audio_player, server.offer, _webrtc_id, "audio").then((connection) => {
pc = connection; pc = connection;
}).catch(() => { }).catch(() => {
console.log("catching") console.info("catching")
dispatch("error", "Too many concurrent users. Come back later!"); dispatch("error", "Too many concurrent users. Come back later!");
}); });
} }
@@ -91,7 +84,6 @@
float={false} float={false}
label={label || i18n("audio.audio")} label={label || i18n("audio.audio")}
/> />
<audio <audio
class="standard-player" class="standard-player"
class:hidden={value === "__webrtc_value__"} class:hidden={value === "__webrtc_value__"}

101
frontend/shared/StaticVideo.svelte
View File

@@ -41,66 +41,59 @@
$: if( value === "start_webrtc_stream") { $: if( value === "start_webrtc_stream") {
value = _webrtc_id; value = _webrtc_id;
const fallback_config = { pc = new RTCPeerConnection(rtc_configuration);
iceServers: [ pc.addEventListener("connectionstatechange",
{ async (event) => {
urls: 'stun:stun.l.google.com:19302' switch(pc.connectionState) {
} case "connected":
] console.log("connected");
}; stream_state = "open";
const configuration = rtc_configuration || fallback_config; break;
pc = new RTCPeerConnection(rtc_configuration); case "disconnected":
pc.addEventListener("connectionstatechange", console.log("closed");
async (event) => { stop(pc);
switch(pc.connectionState) { break;
case "connected": default:
console.log("connected"); break;
stream_state = "open"; }
break; }
case "disconnected": )
console.log("closed"); start(null, pc, video_element, server.offer, _webrtc_id).then((connection) => {
stop(pc); pc = connection;
break; }).catch(() => {
default: console.log("catching")
break; dispatch("error", "Too many concurrent users. Come back later!");
} });
}
)
start(null, pc, video_element, server.offer, _webrtc_id).then((connection) => {
pc = connection;
}).catch(() => {
console.log("catching")
dispatch("error", "Too many concurrent users. Come back later!");
});
} }
</script> </script>
<BlockLabel {show_label} Icon={Video} label={label || "Video"} />
{#if value === "__webrtc_value__"}
<Empty unpadded_box={true} size="large"><Video /></Empty>
{/if}
<div class="wrap"> <div class="wrap">
<BlockLabel {show_label} Icon={Video} label={label || "Video"} /> <video
{#if value === "__webrtc_value__"} class:hidden={value === "__webrtc_value__"}
<Empty unpadded_box={true} size="large"><Video /></Empty> bind:this={video_element}
{/if} autoplay={true}
<video on:loadeddata={dispatch.bind(null, "loadeddata")}
class:hidden={value === "__webrtc_value__"} on:click={dispatch.bind(null, "click")}
bind:this={video_element} on:play={dispatch.bind(null, "play")}
autoplay={true} on:pause={dispatch.bind(null, "pause")}
on:loadeddata={dispatch.bind(null, "loadeddata")} on:ended={dispatch.bind(null, "ended")}
on:click={dispatch.bind(null, "click")} on:mouseover={dispatch.bind(null, "mouseover")}
on:play={dispatch.bind(null, "play")} on:mouseout={dispatch.bind(null, "mouseout")}
on:pause={dispatch.bind(null, "pause")} on:focus={dispatch.bind(null, "focus")}
on:ended={dispatch.bind(null, "ended")} on:blur={dispatch.bind(null, "blur")}
on:mouseover={dispatch.bind(null, "mouseover")} on:load
on:mouseout={dispatch.bind(null, "mouseout")} data-testid={$$props["data-testid"]}
on:focus={dispatch.bind(null, "focus")} crossorigin="anonymous"
on:blur={dispatch.bind(null, "blur")} >
on:load <track kind="captions" />
data-testid={$$props["data-testid"]} </video>
crossorigin="anonymous"
>
<track kind="captions" />
</video>
</div> </div>

18
frontend/shared/Webcam.svelte
View File

@@ -24,7 +24,7 @@
let _time_limit: number | null = null; let _time_limit: number | null = null;
export let time_limit: number | null = null; export let time_limit: number | null = null;
let stream_state: "open" | "waiting" | "closed" = "closed"; let stream_state: "open" | "waiting" | "closed" = "closed";
export const webrtc_id = Math.random().toString(36).substring(2); const _webrtc_id = Math.random().toString(36).substring(2);
export const modify_stream: (state: "open" | "closed" | "waiting") => void = ( export const modify_stream: (state: "open" | "closed" | "waiting") => void = (
state: "open" | "closed" | "waiting" state: "open" | "closed" | "waiting"
@@ -114,19 +114,11 @@
let webcam_accessed = false; let webcam_accessed = false;
let pc: RTCPeerConnection; let pc: RTCPeerConnection;
export let webrtc_id;
async function start_webrtc(): Promise<void> { async function start_webrtc(): Promise<void> {
if (stream_state === 'closed') { if (stream_state === 'closed') {
const fallback_config = { pc = new RTCPeerConnection(rtc_configuration);
iceServers: [
{
urls: 'stun:stun.l.google.com:19302'
}
]
};
const configuration = rtc_configuration || fallback_config;
console.log("config", configuration);
pc = new RTCPeerConnection(configuration);
pc.addEventListener("connectionstatechange", pc.addEventListener("connectionstatechange",
async (event) => { async (event) => {
switch(pc.connectionState) { switch(pc.connectionState) {
@@ -136,6 +128,7 @@
break; break;
case "disconnected": case "disconnected":
stream_state = "closed"; stream_state = "closed";
_time_limit = null;
await access_webcam(); await access_webcam();
break; break;
default: default:
@@ -144,10 +137,11 @@
} }
) )
stream_state = "waiting" stream_state = "waiting"
webrtc_id = _webrtc_id;
start(stream, pc, video_source, server.offer, webrtc_id).then((connection) => { start(stream, pc, video_source, server.offer, webrtc_id).then((connection) => {
pc = connection; pc = connection;
}).catch(() => { }).catch(() => {
console.log("catching") console.info("catching")
stream_state = "closed"; stream_state = "closed";
dispatch("error", "Too many concurrent users. Come back later!"); dispatch("error", "Too many concurrent users. Come back later!");
}); });

29
frontend/shared/webrtc_utils.ts
View File

@@ -3,7 +3,7 @@ export function createPeerConnection(pc, node) {
pc.addEventListener( pc.addEventListener(
"icegatheringstatechange", "icegatheringstatechange",
() => { () => {
console.log(pc.iceGatheringState); console.debug(pc.iceGatheringState);
}, },
false false
); );
@@ -11,7 +11,7 @@ export function createPeerConnection(pc, node) {
pc.addEventListener( pc.addEventListener(
"iceconnectionstatechange", "iceconnectionstatechange",
() => { () => {
console.log(pc.iceConnectionState); console.debug(pc.iceConnectionState);
}, },
false false
); );
@@ -19,25 +19,25 @@ export function createPeerConnection(pc, node) {
pc.addEventListener( pc.addEventListener(
"signalingstatechange", "signalingstatechange",
() => { () => {
console.log(pc.signalingState); console.debug(pc.signalingState);
}, },
false false
); );
// connect audio / video from server to local // connect audio / video from server to local
pc.addEventListener("track", (evt) => { pc.addEventListener("track", (evt) => {
console.log("track event listener"); console.debug("track event listener");
if (node.srcObject !== evt.streams[0]) { if (node.srcObject !== evt.streams[0]) {
console.log("streams", evt.streams); console.debug("streams", evt.streams);
node.srcObject = evt.streams[0]; node.srcObject = evt.streams[0];
console.log("node.srcOject", node.srcObject); console.debug("node.srcOject", node.srcObject);
if (evt.track.kind === 'audio') { if (evt.track.kind === 'audio') {
node.volume = 1.0; // Ensure volume is up node.volume = 1.0; // Ensure volume is up
node.muted = false; node.muted = false;
node.autoplay = true; node.autoplay = true;
// Attempt to play (needed for some browsers) // Attempt to play (needed for some browsers)
node.play().catch(e => console.log("Autoplay failed:", e)); node.play().catch(e => console.debug("Autoplay failed:", e));
} }
} }
}); });
@@ -51,11 +51,11 @@ export async function start(stream, pc: RTCPeerConnection, node, server_fn, webr
stream.getTracks().forEach((track) => { stream.getTracks().forEach((track) => {
track.applyConstraints({ frameRate: { max: 30 } }); track.applyConstraints({ frameRate: { max: 30 } });
console.log("Track stream callback", track); console.debug("Track stream callback", track);
pc.addTrack(track, stream); pc.addTrack(track, stream);
}); });
} else { } else {
console.log("Creating transceiver!"); console.debug("Creating transceiver!");
pc.addTransceiver(modality, { direction: "recvonly" }); pc.addTransceiver(modality, { direction: "recvonly" });
} }
@@ -66,9 +66,9 @@ export async function start(stream, pc: RTCPeerConnection, node, server_fn, webr
function make_offer(server_fn: any, body): Promise<object> { function make_offer(server_fn: any, body): Promise<object> {
return new Promise((resolve, reject) => { return new Promise((resolve, reject) => {
server_fn(body).then((data) => { server_fn(body).then((data) => {
console.log("data", data) console.debug("data", data)
if(data?.status === "failed") { if(data?.status === "failed") {
console.log("rejecting") console.debug("rejecting")
reject("error") reject("error")
} }
resolve(data); resolve(data);
@@ -89,13 +89,13 @@ async function negotiate(
.then(() => { .then(() => {
// wait for ICE gathering to complete // wait for ICE gathering to complete
return new Promise<void>((resolve) => { return new Promise<void>((resolve) => {
console.log("ice gathering state", pc.iceGatheringState); console.debug("ice gathering state", pc.iceGatheringState);
if (pc.iceGatheringState === "complete") { if (pc.iceGatheringState === "complete") {
resolve(); resolve();
} else { } else {
const checkState = () => { const checkState = () => {
if (pc.iceGatheringState === "complete") { if (pc.iceGatheringState === "complete") {
console.log("ice complete"); console.debug("ice complete");
pc.removeEventListener("icegatheringstatechange", checkState); pc.removeEventListener("icegatheringstatechange", checkState);
resolve(); resolve();
} }
@@ -124,8 +124,7 @@ async function negotiate(
} }
export function stop(pc: RTCPeerConnection) { export function stop(pc: RTCPeerConnection) {
console.log("pc", pc); console.debug("Stopping peer connection");
console.log("STOPPING");
// close transceivers // close transceivers
if (pc.getTransceivers) { if (pc.getTransceivers) {
pc.getTransceivers().forEach((transceiver) => { pc.getTransceivers().forEach((transceiver) => {

2
pyproject.toml
View File

@@ -8,7 +8,7 @@ build-backend = "hatchling.build"
[project] [project]
name = "gradio_webrtc" name = "gradio_webrtc"
version = "0.0.1" version = "0.0.2"
description = "Stream images in realtime with webrtc" description = "Stream images in realtime with webrtc"
readme = "README.md" readme = "README.md"
license = "apache-2.0" license = "apache-2.0"