Files
gradio-webrtc/frontend/Index.svelte
neil.xh f476f9cf29 gs对话接入
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
2025-04-16 19:09:04 +08:00

235 lines
7.0 KiB
Svelte

<svelte:options accessors={true} />
<script lang="ts">
import { Block, UploadText } from "@gradio/atoms";
import Video from "./shared/InteractiveVideo.svelte";
import { StatusTracker } from "@gradio/statustracker";
import type { LoadingStatus } from "@gradio/statustracker";
import StaticVideo from "./shared/StaticVideo.svelte";
import StaticAudio from "./shared/StaticAudio.svelte";
import InteractiveAudio from "./shared/InteractiveAudio.svelte";
import VideoChat from './shared/VideoChat/index.svelte'
export let video_chat = false;
export let avatar_type: string = ""
export let avatar_ws_route: string = ""
export let avatar_assets_path: string = ""
export let elem_id = "";
export let elem_classes: string[] = [];
export let visible = true;
export let value: string = "__webrtc_value__";
export let button_labels: { start: string; stop: string; waiting: string };
export let label: string;
export let root: string;
export let show_label: boolean;
export let loading_status: LoadingStatus;
export let height: number | undefined;
export let width: number | undefined;
export let server: {
offer: (body: any) => Promise<any>;
};
export let container = false;
export let scale: number | null = null;
export let min_width: number | undefined = undefined;
export let gradio;
export let rtc_configuration: Object;
export let time_limit: number | null = null;
export let modality: "video" | "audio" | "audio-video" = "video";
export let mode: "send-receive" | "receive" | "send" = "send-receive";
export let rtp_params: RTCRtpParameters = {} as RTCRtpParameters;
export let track_constraints: MediaTrackConstraints = {};
export let icon: string | undefined = undefined;
export let icon_button_color: string = "var(--color-accent)";
export let pulse_color: string = "var(--color-accent)";
export let icon_radius: number = 50;
const on_change_cb = (msg: "change" | "tick" | any) => {
if (
msg?.type === "info" ||
msg?.type === "warning" ||
msg?.type === "error"
) {
gradio.dispatch(msg?.type === "error" ? "error" : "warning", msg.message);
} else if (msg?.type === "end_stream") {
gradio.dispatch("warning", msg.data);
} else if (msg?.type === "fetch_output") {
gradio.dispatch("state_change");
} else if (msg?.type === "send_input") {
gradio.dispatch("tick");
} else if (msg?.type === "connection_timeout") {
gradio.dispatch(
"warning",
"Taking a while to connect. Are you on a VPN?",
);
}
if (msg.type === "state_change") {
gradio.dispatch(msg === "change" ? "state_change" : "tick");
}
};
const reject_cb = (msg: object) => {
if (
msg.status === "failed" &&
msg.meta?.error === "concurrency_limit_reached"
) {
gradio.dispatch(
"error",
`Too many concurrent connections. Please try again later!`,
);
} else {
gradio.dispatch("error", "Unexpected server error");
}
};
let dragging = false;
</script>
{#if video_chat}
<Block
{visible}
variant={"solid"}
border_mode={dragging ? "focus" : "base"}
padding={false}
{elem_id}
{elem_classes}
{height}
{width}
{container}
{scale}
{min_width}
allow_overflow={false}>
<VideoChat {server} bind:webrtc_id={value}
on:clear={() => gradio.dispatch("clear")}
on:play={() => gradio.dispatch("play")}
on:pause={() => gradio.dispatch("pause")}
on:upload={() => gradio.dispatch("upload")}
on:stop={() => gradio.dispatch("stop")}
on:end={() => gradio.dispatch("end")}
on:start_recording={() => gradio.dispatch("start_recording")}
on:stop_recording={() => gradio.dispatch("stop_recording")}
on:tick={() => gradio.dispatch("tick")}
on:error={({ detail }) => gradio.dispatch("error", detail)}
i18n={gradio.i18n}
stream_handler={(...args) => gradio.client.stream(...args)}
{avatar_type}
{avatar_ws_route}
{avatar_assets_path}
{track_constraints}
{height}
{on_change_cb} {rtc_configuration}
on:tick={() => gradio.dispatch("tick")}
on:error={({ detail }) => gradio.dispatch("error", detail)}></VideoChat>
</Block>
{:else}
<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)}
/>
{#if mode == "receive" && modality === "video"}
<StaticVideo
bind:value
{on_change_cb}
{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
{on_change_cb}
{label}
{show_label}
{server}
{rtc_configuration}
{icon}
{icon_button_color}
{pulse_color}
{icon_radius}
i18n={gradio.i18n}
on:tick={() => gradio.dispatch("tick")}
on:error={({ detail }) => gradio.dispatch("error", detail)}
/>
{:else if (mode === "send-receive" || mode == "send") && (modality === "video" || modality == "audio-video")}
<Video
bind:value
{label}
{show_label}
active_source={"webcam"}
include_audio={modality === "audio-video"}
{server}
{rtc_configuration}
{time_limit}
{mode}
{track_constraints}
{rtp_params}
{on_change_cb}
{reject_cb}
{icon}
{icon_button_color}
{pulse_color}
{icon_radius}
{button_labels}
on:clear={() => gradio.dispatch("clear")}
on:play={() => gradio.dispatch("play")}
on:pause={() => gradio.dispatch("pause")}
on:upload={() => gradio.dispatch("upload")}
on:stop={() => gradio.dispatch("stop")}
on:end={() => gradio.dispatch("end")}
on:start_recording={() => gradio.dispatch("start_recording")}
on:stop_recording={() => gradio.dispatch("stop_recording")}
on:tick={() => gradio.dispatch("tick")}
on:error={({ detail }) => gradio.dispatch("error", detail)}
i18n={gradio.i18n}
stream_handler={(...args) => gradio.client.stream(...args)}
>
<UploadText i18n={gradio.i18n} type="video" />
</Video>
{:else if (mode === "send-receive" || mode === "send") && modality === "audio"}
<InteractiveAudio
bind:value
{on_change_cb}
{label}
{show_label}
{server}
{rtc_configuration}
{time_limit}
{track_constraints}
{mode}
{rtp_params}
i18n={gradio.i18n}
{icon}
{reject_cb}
{icon_button_color}
{icon_radius}
{pulse_color}
{button_labels}
on:tick={() => gradio.dispatch("tick")}
on:error={({ detail }) => gradio.dispatch("error", detail)}
on:warning={({ detail }) => gradio.dispatch("warning", detail)}
/>
{/if}
</Block>
{/if}