mirror of
https://github.com/HumanAIGC-Engineering/gradio-webrtc.git
synced 2026-02-04 09:29:23 +08:00
本次代码评审新增并完善了gs视频聊天功能,包括前后端接口定义、状态管理及UI组件实现,并引入了新的依赖库以支持更多互动特性。 Link: https://code.alibaba-inc.com/xr-paas/gradio_webrtc/codereview/21273476 * 更新python 部分 * 合并videochat前端部分 * Merge branch 'feature/update-fastrtc-0.0.19' of http://gitlab.alibaba-inc.com/xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19 * 替换audiowave * 导入路径修改 * 合并websocket mode逻辑 * feat: gaussian avatar chat * 增加其他渲染的入参 * feat: ws连接和使用 * Merge branch 'feature/update-fastrtc-0.0.19' of http://gitlab.alibaba-inc.com/xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19 * 右边距离超出容器宽度,则向左移动 * 配置传递 * Merge branch 'feature/update-fastrtc-0.0.19' of gitlab.alibaba-inc.com:xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19 * 高斯包异常 * 同步webrtc_utils * 更新webrtc_utils * 兼容on_chat_datachannel * 修复设备名称列表没有正常显示的问题 * copy 传递 webrtc_id * Merge branch 'feature/update-fastrtc-0.0.19' of gitlab.alibaba-inc.com:xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19 * 保证webrtc 完成后再进行websocket连接 * feat: 音频表情数据接入 * dist 上传 * canvas 隐藏 * feat: 高斯文件下载进度透出 * Merge branch 'feature/update-fastrtc-0.0.19' of http://gitlab.alibaba-inc.com/xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19 * 修改无法获取权限问题 * Merge branch 'feature/update-fastrtc-0.0.19' of gitlab.alibaba-inc.com:xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19 * 先获取权限再获取设备 * fix: gs资源下载完成前不处理ws数据 * fix: merge * 话术调整 * Merge branch 'feature/update-fastrtc-0.0.19' of gitlab.alibaba-inc.com:xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19 * 修复设备切换后重新对话,又切换回默认设备的问题 * Merge branch 'feature/update-fastrtc-0.0.19' of http://gitlab.alibaba-inc.com/xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19 * 更新localvideo 尺寸 * Merge branch 'feature/update-fastrtc-0.0.19' of gitlab.alibaba-inc.com:xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19 * 不能默认default * 修改音频权限问题 * 更新打包结果 * fix: 对话按钮状态跟gs资源挂钩,删除无用代码 * fix: merge * feat: gs渲染模块从npm包引入 * fix * 新增对话记录 * Merge branch 'feature/update-fastrtc-0.0.19' of http://gitlab.alibaba-inc.com/xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19 * 样式修改 * 更新包 * fix: gs数字人初始化位置和静音 * 对话记录滚到底部 * 至少100%高度 * Merge branch 'feature/update-fastrtc-0.0.19' of gitlab.alibaba-inc.com:xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19 * 略微上移文本框 * 开始连接时清空对话记录 * fix: update gs render npm * Merge branch 'feature/update-fastrtc-0.0.19' of http://gitlab.alibaba-inc.com/xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19 * 逻辑保证 * Merge branch 'feature/update-fastrtc-0.0.19' of gitlab.alibaba-inc.com:xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19 * feat: 音频初始化配置是否静音 * actionsbar在有字幕时调整位置 * Merge branch 'feature/update-fastrtc-0.0.19' of http://gitlab.alibaba-inc.com/xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19 * 样式优化 * feat: 增加readme * fix: 资源图片 * fix: docs * fix: update gs render sdk * fix: gs模式下画面位置计算 * fix: update readme * 设备判断,太窄处理 * Merge branch 'feature/update-fastrtc-0.0.19' of gitlab.alibaba-inc.com:xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19 * 是否有权限和是否有设备分开 * feat: gs 下载和加载钩子函数分离 * Merge branch 'feature/update-fastrtc-0.0.19' of http://gitlab.alibaba-inc.com/xr-paas/gradio_webrtc into feature/update-fastrtc-0.0.19 * fix: update gs render sdk * 替换 * dist * 上传文件 * del
235 lines
7.0 KiB
Svelte
235 lines
7.0 KiB
Svelte
<svelte:options accessors={true} />
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<script lang="ts">
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import { Block, UploadText } from "@gradio/atoms";
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import Video from "./shared/InteractiveVideo.svelte";
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import { StatusTracker } from "@gradio/statustracker";
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import type { LoadingStatus } from "@gradio/statustracker";
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import StaticVideo from "./shared/StaticVideo.svelte";
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import StaticAudio from "./shared/StaticAudio.svelte";
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import InteractiveAudio from "./shared/InteractiveAudio.svelte";
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import VideoChat from './shared/VideoChat/index.svelte'
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export let video_chat = false;
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export let avatar_type: string = ""
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export let avatar_ws_route: string = ""
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export let avatar_assets_path: string = ""
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export let elem_id = "";
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export let elem_classes: string[] = [];
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export let visible = true;
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export let value: string = "__webrtc_value__";
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export let button_labels: { start: string; stop: string; waiting: string };
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export let label: string;
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export let root: string;
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export let show_label: boolean;
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export let loading_status: LoadingStatus;
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export let height: number | undefined;
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export let width: number | undefined;
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export let server: {
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offer: (body: any) => Promise<any>;
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};
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export let container = false;
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export let scale: number | null = null;
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export let min_width: number | undefined = undefined;
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export let gradio;
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export let rtc_configuration: Object;
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export let time_limit: number | null = null;
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export let modality: "video" | "audio" | "audio-video" = "video";
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export let mode: "send-receive" | "receive" | "send" = "send-receive";
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export let rtp_params: RTCRtpParameters = {} as RTCRtpParameters;
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export let track_constraints: MediaTrackConstraints = {};
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export let icon: string | undefined = undefined;
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export let icon_button_color: string = "var(--color-accent)";
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export let pulse_color: string = "var(--color-accent)";
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export let icon_radius: number = 50;
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const on_change_cb = (msg: "change" | "tick" | any) => {
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if (
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msg?.type === "info" ||
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msg?.type === "warning" ||
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msg?.type === "error"
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) {
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gradio.dispatch(msg?.type === "error" ? "error" : "warning", msg.message);
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} else if (msg?.type === "end_stream") {
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gradio.dispatch("warning", msg.data);
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} else if (msg?.type === "fetch_output") {
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gradio.dispatch("state_change");
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} else if (msg?.type === "send_input") {
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gradio.dispatch("tick");
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} else if (msg?.type === "connection_timeout") {
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gradio.dispatch(
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"warning",
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"Taking a while to connect. Are you on a VPN?",
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);
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}
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if (msg.type === "state_change") {
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gradio.dispatch(msg === "change" ? "state_change" : "tick");
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}
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};
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const reject_cb = (msg: object) => {
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if (
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msg.status === "failed" &&
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msg.meta?.error === "concurrency_limit_reached"
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) {
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gradio.dispatch(
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"error",
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`Too many concurrent connections. Please try again later!`,
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);
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} else {
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gradio.dispatch("error", "Unexpected server error");
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}
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};
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let dragging = false;
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</script>
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{#if video_chat}
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<Block
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{visible}
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variant={"solid"}
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border_mode={dragging ? "focus" : "base"}
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padding={false}
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{elem_id}
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{elem_classes}
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{height}
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{width}
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{container}
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{scale}
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{min_width}
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allow_overflow={false}>
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<VideoChat {server} bind:webrtc_id={value}
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on:clear={() => gradio.dispatch("clear")}
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on:play={() => gradio.dispatch("play")}
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on:pause={() => gradio.dispatch("pause")}
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on:upload={() => gradio.dispatch("upload")}
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on:stop={() => gradio.dispatch("stop")}
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on:end={() => gradio.dispatch("end")}
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on:start_recording={() => gradio.dispatch("start_recording")}
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on:stop_recording={() => gradio.dispatch("stop_recording")}
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on:tick={() => gradio.dispatch("tick")}
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on:error={({ detail }) => gradio.dispatch("error", detail)}
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i18n={gradio.i18n}
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stream_handler={(...args) => gradio.client.stream(...args)}
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{avatar_type}
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{avatar_ws_route}
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{avatar_assets_path}
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{track_constraints}
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{height}
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{on_change_cb} {rtc_configuration}
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on:tick={() => gradio.dispatch("tick")}
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on:error={({ detail }) => gradio.dispatch("error", detail)}></VideoChat>
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</Block>
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{:else}
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<Block
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{visible}
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variant={"solid"}
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border_mode={dragging ? "focus" : "base"}
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padding={false}
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{elem_id}
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{elem_classes}
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{height}
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{width}
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{container}
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{scale}
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{min_width}
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allow_overflow={false}
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>
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<StatusTracker
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autoscroll={gradio.autoscroll}
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i18n={gradio.i18n}
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{...loading_status}
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on:clear_status={() => gradio.dispatch("clear_status", loading_status)}
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/>
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{#if mode == "receive" && modality === "video"}
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<StaticVideo
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bind:value
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{on_change_cb}
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{label}
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{show_label}
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{server}
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{rtc_configuration}
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on:tick={() => gradio.dispatch("tick")}
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on:error={({ detail }) => gradio.dispatch("error", detail)}
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/>
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{:else if mode == "receive" && modality === "audio"}
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<StaticAudio
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bind:value
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{on_change_cb}
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{label}
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{show_label}
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{server}
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{rtc_configuration}
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{icon}
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{icon_button_color}
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{pulse_color}
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{icon_radius}
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i18n={gradio.i18n}
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on:tick={() => gradio.dispatch("tick")}
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on:error={({ detail }) => gradio.dispatch("error", detail)}
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/>
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{:else if (mode === "send-receive" || mode == "send") && (modality === "video" || modality == "audio-video")}
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<Video
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bind:value
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{label}
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{show_label}
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active_source={"webcam"}
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include_audio={modality === "audio-video"}
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{server}
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{rtc_configuration}
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{time_limit}
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{mode}
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{track_constraints}
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{rtp_params}
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{on_change_cb}
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{reject_cb}
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{icon}
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{icon_button_color}
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{pulse_color}
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{icon_radius}
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{button_labels}
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on:clear={() => gradio.dispatch("clear")}
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on:play={() => gradio.dispatch("play")}
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on:pause={() => gradio.dispatch("pause")}
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on:upload={() => gradio.dispatch("upload")}
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on:stop={() => gradio.dispatch("stop")}
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on:end={() => gradio.dispatch("end")}
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on:start_recording={() => gradio.dispatch("start_recording")}
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on:stop_recording={() => gradio.dispatch("stop_recording")}
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on:tick={() => gradio.dispatch("tick")}
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on:error={({ detail }) => gradio.dispatch("error", detail)}
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i18n={gradio.i18n}
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stream_handler={(...args) => gradio.client.stream(...args)}
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>
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<UploadText i18n={gradio.i18n} type="video" />
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</Video>
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{:else if (mode === "send-receive" || mode === "send") && modality === "audio"}
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<InteractiveAudio
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bind:value
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{on_change_cb}
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{label}
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{show_label}
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{server}
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{rtc_configuration}
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{time_limit}
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{track_constraints}
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{mode}
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{rtp_params}
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i18n={gradio.i18n}
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{icon}
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{reject_cb}
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{icon_button_color}
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{icon_radius}
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{pulse_color}
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{button_labels}
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on:tick={() => gradio.dispatch("tick")}
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on:error={({ detail }) => gradio.dispatch("error", detail)}
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on:warning={({ detail }) => gradio.dispatch("warning", detail)}
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/>
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{/if}
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</Block>
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{/if} |