mirror of
https://github.com/HumanAIGC-Engineering/gradio-webrtc.git
synced 2026-02-05 01:49:23 +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
194 lines
4.9 KiB
TypeScript
194 lines
4.9 KiB
TypeScript
export function createPeerConnection(pc, node) {
|
|
// register some listeners to help debugging
|
|
pc.addEventListener(
|
|
"icegatheringstatechange",
|
|
() => {
|
|
console.debug(pc.iceGatheringState);
|
|
},
|
|
false,
|
|
);
|
|
|
|
pc.addEventListener(
|
|
"iceconnectionstatechange",
|
|
() => {
|
|
console.debug(pc.iceConnectionState);
|
|
},
|
|
false,
|
|
);
|
|
|
|
pc.addEventListener(
|
|
"signalingstatechange",
|
|
() => {
|
|
console.debug(pc.signalingState);
|
|
},
|
|
false,
|
|
);
|
|
|
|
// connect audio / video from server to local
|
|
pc.addEventListener("track", (evt) => {
|
|
console.debug("track event listener");
|
|
if (node && node.srcObject !== evt.streams[0]) {
|
|
console.debug("streams", evt.streams);
|
|
node.srcObject = evt.streams[0];
|
|
console.debug("node.srcOject", node.srcObject);
|
|
if (evt.track.kind === "audio") {
|
|
node.volume = 1.0; // Ensure volume is up
|
|
node.muted = false;
|
|
node.autoplay = true;
|
|
// Attempt to play (needed for some browsers)
|
|
node.play().catch((e) => console.debug("Autoplay failed:", e));
|
|
}
|
|
}
|
|
});
|
|
|
|
return pc;
|
|
}
|
|
|
|
export async function start(
|
|
stream,
|
|
pc: RTCPeerConnection,
|
|
node,
|
|
server_fn,
|
|
webrtc_id,
|
|
modality: "video" | "audio" = "video",
|
|
on_change_cb: (msg: "change" | "tick") => void = () => {},
|
|
rtp_params = {},
|
|
additional_message_cb: (msg: object) => void = () => {},
|
|
reject_cb: (msg: object) => void = () => {},
|
|
) {
|
|
pc = createPeerConnection(pc, node);
|
|
const data_channel = pc.createDataChannel("text");
|
|
|
|
data_channel.onopen = () => {
|
|
console.debug("Data channel is open");
|
|
data_channel.send("handshake");
|
|
data_channel.send(JSON.stringify({type:'init'}));
|
|
};
|
|
|
|
data_channel.onmessage = (event) => {
|
|
console.debug("Received message:", event.data);
|
|
let event_json;
|
|
try {
|
|
event_json = JSON.parse(event.data);
|
|
} catch (e) {
|
|
console.debug("Error parsing JSON");
|
|
}
|
|
if (
|
|
event.data === "change" ||
|
|
event.data === "tick" ||
|
|
event.data === "stopword" ||
|
|
event_json?.type === "warning" ||
|
|
event_json?.type === "error" ||
|
|
event_json?.type === "send_input" ||
|
|
event_json?.type === "fetch_output" ||
|
|
event_json?.type === "stopword" ||
|
|
event_json?.type === "end_stream"
|
|
) {
|
|
on_change_cb(event_json ?? event.data);
|
|
}
|
|
additional_message_cb(event_json ?? event.data);
|
|
};
|
|
|
|
if (stream) {
|
|
stream.getTracks().forEach(async (track) => {
|
|
console.debug("Track stream callback", track);
|
|
const sender = pc.addTrack(track, stream);
|
|
const params = sender.getParameters();
|
|
const updated_params = { ...params, ...rtp_params };
|
|
await sender.setParameters(updated_params);
|
|
console.debug("sender params", sender.getParameters());
|
|
});
|
|
} else {
|
|
console.debug("Creating transceiver!");
|
|
pc.addTransceiver(modality, { direction: "recvonly" });
|
|
}
|
|
|
|
await negotiate(pc, server_fn, webrtc_id, reject_cb);
|
|
return [pc, data_channel] as const;
|
|
}
|
|
|
|
function make_offer(
|
|
server_fn: any,
|
|
body,
|
|
reject_cb: (msg: object) => void = () => {},
|
|
): Promise<object> {
|
|
return new Promise((resolve, reject) => {
|
|
server_fn(body).then((data) => {
|
|
console.debug("data", data);
|
|
if (data?.status === "failed") {
|
|
reject_cb(data);
|
|
console.debug("rejecting");
|
|
reject("error");
|
|
}
|
|
resolve(data);
|
|
});
|
|
});
|
|
}
|
|
|
|
async function negotiate(
|
|
pc: RTCPeerConnection,
|
|
server_fn: any,
|
|
webrtc_id: string,
|
|
reject_cb: (msg: object) => void = () => {},
|
|
): Promise<void> {
|
|
pc.onicecandidate = ({ candidate }) => {
|
|
if (candidate) {
|
|
console.debug("Sending ICE candidate", candidate);
|
|
server_fn({
|
|
candidate: candidate.toJSON(),
|
|
webrtc_id: webrtc_id,
|
|
type: "ice-candidate",
|
|
}).catch((err) => console.error("Error sending ICE candidate:", err));
|
|
}
|
|
};
|
|
|
|
return pc
|
|
.createOffer()
|
|
.then((offer) => {
|
|
return pc.setLocalDescription(offer);
|
|
})
|
|
.then(() => {
|
|
var offer = pc.localDescription;
|
|
return make_offer(
|
|
server_fn,
|
|
{
|
|
sdp: offer.sdp,
|
|
type: offer.type,
|
|
webrtc_id: webrtc_id,
|
|
},
|
|
reject_cb,
|
|
);
|
|
})
|
|
.then((response) => {
|
|
return response;
|
|
})
|
|
.then((answer) => {
|
|
return pc.setRemoteDescription(answer);
|
|
});
|
|
}
|
|
|
|
export function stop(pc: RTCPeerConnection) {
|
|
console.debug("Stopping peer connection");
|
|
// close transceivers
|
|
if (pc.getTransceivers) {
|
|
pc.getTransceivers().forEach((transceiver) => {
|
|
if (transceiver.stop) {
|
|
transceiver.stop();
|
|
}
|
|
});
|
|
}
|
|
|
|
// close local audio / video
|
|
if (pc.getSenders()) {
|
|
pc.getSenders().forEach((sender) => {
|
|
console.log("sender", sender);
|
|
if (sender.track && sender.track.stop) sender.track.stop();
|
|
});
|
|
}
|
|
|
|
// close peer connection
|
|
setTimeout(() => {
|
|
pc.close();
|
|
}, 500);
|
|
}
|