refactor audio calls into manager class and rewrite user interface
This commit is contained in:
liamcottle
452
public/call.html
452
public/call.html
@@ -1,73 +1,262 @@
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<html>
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<body>
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<div>
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<head>
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<div>
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<button onclick="startStreaming()">Start Streaming</button>
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<button onclick="stopStreaming()">Stop Streaming</button>
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<button onclick="startListening()">Start Listening</button>
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<button onclick="stopListening()">Stop Listening</button>
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<meta charset="UTF-8">
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<meta http-equiv="X-UA-Compatible" content="IE=edge">
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<meta name="viewport" content="width=device-width, initial-scale=1, maximum-scale=1">
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<title>Phone | Reticulum WebChat</title>
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<!-- scripts -->
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<script src="assets/js/tailwindcss/tailwind-v3.4.3-forms-v0.5.7.js"></script>
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<script src="assets/js/vue@3.4.26/dist/vue.global.js"></script>
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<!-- codec2 -->
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<script src="assets/js/codec2-emscripten/c2enc.js"></script>
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<script src="assets/js/codec2-emscripten/c2dec.js"></script>
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<script src="assets/js/codec2-emscripten/sox.js"></script>
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<script src="assets/js/codec2-emscripten/codec2-lib.js"></script>
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</head>
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<body class="bg-gray-100">
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<div id="app" class="flex h-full">
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<div class="mx-auto my-auto w-full max-w-md">
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<!-- in active call -->
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<div v-if="isWebsocketConnected" class="w-full">
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<div class="border rounded-xl bg-white shadow w-full">
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<div class="flex border-b border-gray-300 text-gray-700 p-2">
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<div class="my-auto mr-2">
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<svg xmlns="http://www.w3.org/2000/svg" fill="none" viewBox="0 0 24 24" stroke-width="1.5" stroke="currentColor" class="w-6 h-6">
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<path stroke-linecap="round" stroke-linejoin="round" d="M2.25 6.75c0 8.284 6.716 15 15 15h2.25a2.25 2.25 0 0 0 2.25-2.25v-1.372c0-.516-.351-.966-.852-1.091l-4.423-1.106c-.44-.11-.902.055-1.173.417l-.97 1.293c-.282.376-.769.542-1.21.38a12.035 12.035 0 0 1-7.143-7.143c-.162-.441.004-.928.38-1.21l1.293-.97c.363-.271.527-.734.417-1.173L6.963 3.102a1.125 1.125 0 0 0-1.091-.852H4.5A2.25 2.25 0 0 0 2.25 4.5v2.25Z" />
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</svg>
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</div>
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<div class="my-auto">Reticulum Phone</div>
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</div>
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<div class="border-b border-gray-300 text-gray-700 p-2">
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<div class="mb-2">
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<div class="mb-1 text-sm font-medium text-gray-900">Call Hash</div>
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<div class="text-xs text-gray-600">{{ callHash }}</div>
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</div>
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<div class="mb-2">
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<div class="mb-1 text-sm font-medium text-gray-900">TX Bytes</div>
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<div class="text-xs text-gray-600">{{ formatBytes(txBytes) }}</div>
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</div>
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<div class="mb-2">
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<div class="mb-1 text-sm font-medium text-gray-900">RX Bytes</div>
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<div class="text-xs text-gray-600">{{ formatBytes(rxBytes) }}</div>
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</div>
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<div>
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<input id="checkbox-mute-mic" type="checkbox"/> Mute Mic
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</div>
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<div>
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<select id="codec-mode">
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<div class="mb-1 text-sm font-medium text-gray-900">Codec2 Mode</div>
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<select v-model="codecMode" class="bg-gray-50 border border-gray-300 text-gray-900 text-sm rounded-lg focus:ring-blue-500 focus:border-blue-500 block w-full p-2.5">
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<option value="3200">3200</option>
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<option value="2400">2400</option>
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<option value="1600">1600</option>
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<option value="1400">1400</option>
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<option value="1300">1300</option>
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<option value="1200" selected>1200</option>
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<option value="1200">1200</option>
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<option value="700C">700C</option>
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<option value="450">450</option>
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<option value="450PWB">450PWB</option>
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</select>
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</div>
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<div>Encoded Bytes Sent: <span id="encoded-bytes-sent"></span></div>
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</div>
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<div class="flex text-gray-900 p-2">
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<button @click="isMicMuted = !isMicMuted" type="button" :class="[ isMicMuted ? 'bg-red-500 hover:bg-red-400 focus-visible:outline-red-500' : 'bg-green-500 hover:bg-green-400 focus-visible:outline-green-500' ]" class="my-auto inline-flex items-center gap-x-1 rounded-md p-2 text-sm font-semibold text-white shadow-sm focus-visible:outline focus-visible:outline-2 focus-visible:outline-offset-2">
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<svg xmlns="http://www.w3.org/2000/svg" viewBox="0 0 20 20" fill="currentColor" class="w-5 h-5">
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<path d="M7 4a3 3 0 0 1 6 0v6a3 3 0 1 1-6 0V4Z" />
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<path d="M5.5 9.643a.75.75 0 0 0-1.5 0V10c0 3.06 2.29 5.585 5.25 5.954V17.5h-1.5a.75.75 0 0 0 0 1.5h4.5a.75.75 0 0 0 0-1.5h-1.5v-1.546A6.001 6.001 0 0 0 16 10v-.357a.75.75 0 0 0-1.5 0V10a4.5 4.5 0 0 1-9 0v-.357Z" />
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</svg>
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</button>
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<button @click="leaveCall" type="button" class="ml-auto my-auto inline-flex items-center gap-x-1 rounded-md bg-red-500 p-2 text-sm font-semibold text-white shadow-sm hover:bg-red-400 focus-visible:outline focus-visible:outline-2 focus-visible:outline-offset-2 focus-visible:outline-red-500">
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Leave Call
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</button>
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</div>
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</div>
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</div>
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<!-- no call active -->
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<div v-else class="w-full">
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<div class="border rounded-xl bg-white shadow w-full">
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<div class="flex border-b border-gray-300 text-gray-700 p-2">
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<div class="my-auto mr-2">
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<svg xmlns="http://www.w3.org/2000/svg" fill="none" viewBox="0 0 24 24" stroke-width="1.5" stroke="currentColor" class="w-6 h-6">
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<path stroke-linecap="round" stroke-linejoin="round" d="M2.25 6.75c0 8.284 6.716 15 15 15h2.25a2.25 2.25 0 0 0 2.25-2.25v-1.372c0-.516-.351-.966-.852-1.091l-4.423-1.106c-.44-.11-.902.055-1.173.417l-.97 1.293c-.282.376-.769.542-1.21.38a12.035 12.035 0 0 1-7.143-7.143c-.162-.441.004-.928.38-1.21l1.293-.97c.363-.271.527-.734.417-1.173L6.963 3.102a1.125 1.125 0 0 0-1.091-.852H4.5A2.25 2.25 0 0 0 2.25 4.5v2.25Z" />
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</svg>
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</div>
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<div class="my-auto">Reticulum Phone</div>
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</div>
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<div class="flex text-gray-900 p-2 space-x-2">
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<div class="flex-1">
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<input v-model="callHash" type="text" placeholder="Enter Call Hash" class="bg-gray-50 border border-gray-300 text-gray-900 text-sm rounded-lg focus:ring-blue-500 focus:border-blue-500 block w-full p-2">
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</div>
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<button @click="joinCall" type="button" class="my-auto inline-flex items-center gap-x-1 rounded-md bg-gray-500 p-2 text-sm font-semibold text-white shadow-sm hover:bg-gray-400 focus-visible:outline focus-visible:outline-2 focus-visible:outline-offset-2 focus-visible:outline-gray-500">
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Join Existing Call
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</button>
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</div>
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</div>
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</div>
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</div>
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<script src="assets/js/codec2-emscripten/c2enc.js"></script>
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<script src="assets/js/codec2-emscripten/c2dec.js"></script>
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<script src="assets/js/codec2-emscripten/sox.js"></script>
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<script src="assets/js/codec2-emscripten/codec2-lib.js"></script>
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</div>
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<script>
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Vue.createApp({
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data() {
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return {
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// find elements
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const codecModeElement = document.getElementById("codec-mode");
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const encodedBytesSentElement = document.getElementById("encoded-bytes-sent");
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const checkboxMuteMicElement = document.getElementById("checkbox-mute-mic");
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isWebsocketConnected: false,
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callHash: null,
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txBytes: 0,
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rxBytes: 0,
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var encodedBytesSent = 0;
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isMicMuted: true,
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codecMode: "1200",
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sampleRate: 8000,
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let audioContext;
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let mediaStreamSource;
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let audioWorkletNode;
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let microphoneMediaStream;
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audioContext: null,
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mediaStreamSource: null,
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audioWorkletNode: null,
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microphoneMediaStream: null,
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var callWebsocket = null;
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var listenWebsocket = null;
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};
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},
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mounted: function() {
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async function startRecordingMicrophone(onAudioAvailable) {
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},
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methods: {
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async joinCall() {
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// make sure call hash provided
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if(!this.callHash){
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alert("Enter hash of existing call.");
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return;
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}
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// reset stats
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this.txBytes = 0;
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this.rxBytes = 0;
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// connect to websocket
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this.ws = new WebSocket(location.origin.replace(/^http/, 'ws') + `/api/v1/calls/${this.callHash}/audio`);
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this.ws.addEventListener('open', async () => {
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// we are now connected
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this.isWebsocketConnected = true;
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// send mic audio over call
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await this.startRecordingMicrophone((encoded) => {
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// do nothing if websocket closed
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if(this.ws.readyState !== WebSocket.OPEN){
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return;
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}
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// do nothing when audio muted
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if(this.isMicMuted){
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return;
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}
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// send encoded audio to websocket
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this.ws.send(encoded);
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// update stats
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this.txBytes += encoded.length;
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});
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});
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this.ws.addEventListener('close', () => {
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this.isWebsocketConnected = false;
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this.leaveCall();
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});
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this.ws.addEventListener('error', (error) => {
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console.log(error);
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});
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// listen to audio from call
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this.ws.onmessage = async (event) => {
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// get encoded codec2 bytes from websocket message
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const encoded = await event.data.arrayBuffer();
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// update stats
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this.rxBytes += encoded.byteLength;
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// decode codec2 audio
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const decoded = await Codec2Lib.runDecode(this.codecMode, encoded);
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// convert decoded codec2 to wav audio
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const wavAudio = await Codec2Lib.rawToWav(decoded);
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// play wav audio buffer
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let audioCtx = new AudioContext()
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const audioBuffer = await audioCtx.decodeAudioData(wavAudio.buffer);
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const sampleSource = audioCtx.createBufferSource();
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sampleSource.buffer = audioBuffer;
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sampleSource.connect(audioCtx.destination)
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sampleSource.start(0);
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};
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},
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leaveCall: function() {
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// mute mic
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this.isMicMuted = true;
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// disconnect websocket
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if(this.ws){
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this.ws.close();
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}
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// disconnect media stream source
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if(this.mediaStreamSource){
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this.mediaStreamSource.disconnect();
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}
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// stop using microphone
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if(this.microphoneMediaStream){
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this.microphoneMediaStream.getTracks().forEach(track => track.stop());
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}
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// disconnect the audio worklet node
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if(this.audioWorkletNode){
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this.audioWorkletNode.disconnect();
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}
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// close audio context
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if(this.audioContext && this.audioContext.state !== "closed"){
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this.audioContext.close();
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}
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},
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async startRecordingMicrophone(onAudioAvailable) {
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try {
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// load audio worklet module
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audioContext = new AudioContext({ sampleRate: 8000 });
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await audioContext.audioWorklet.addModule('assets/js/codec2-emscripten/processor.js');
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audioWorkletNode = new AudioWorkletNode(audioContext, 'audio-processor');
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this.audioContext = new AudioContext({ sampleRate: this.sampleRate });
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await this.audioContext.audioWorklet.addModule('assets/js/codec2-emscripten/processor.js');
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this.audioWorkletNode = new AudioWorkletNode(this.audioContext, 'audio-processor');
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// handle audio received from audio worklet
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audioWorkletNode.port.onmessage = async (event) => {
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this.audioWorkletNode.port.onmessage = async (event) => {
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// convert audio received from worklet processor to wav
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const buffer = encodeWAV(event.data, 8000);
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const buffer = this.encodeWAV(event.data, this.sampleRate);
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// convert wav audio to codec2
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const rawBuffer = await Codec2Lib.audioFileToRaw(buffer, "audio.wav");
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const encoded = await Codec2Lib.runEncode(codecModeElement.value, rawBuffer);
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const encoded = await Codec2Lib.runEncode(this.codecMode, rawBuffer);
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// pass encoded audio to callback
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onAudioAvailable(encoded);
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@@ -75,178 +264,31 @@
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};
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// request access to the microphone
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microphoneMediaStream = await navigator.mediaDevices.getUserMedia({ audio: true });
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this.microphoneMediaStream = await navigator.mediaDevices.getUserMedia({
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audio: true,
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});
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// send mic audio to audio worklet
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mediaStreamSource = audioContext.createMediaStreamSource(microphoneMediaStream);
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mediaStreamSource.connect(audioWorkletNode);
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this.mediaStreamSource = this.audioContext.createMediaStreamSource(this.microphoneMediaStream);
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this.mediaStreamSource.connect(this.audioWorkletNode);
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} catch(e) {
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alert(e);
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console.log(e);
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}
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}
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async function startStreaming() {
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try {
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// reset stats
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encodedBytesSent = 0;
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// ask who to call
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const destinationHash = prompt("Enter destination hash to call");
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if(!destinationHash){
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return;
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}
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// connect to websocket
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callWebsocket = new WebSocket(location.origin.replace(/^http/, 'ws') + "/call/initiate/" + destinationHash);
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callWebsocket.onmessage = async function(event) {
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// get encoded codec2 bytes from websocket message
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const encoded = await event.data.arrayBuffer();
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// decode codec2 audio
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const decoded = await Codec2Lib.runDecode(codecModeElement.value, encoded);
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// convert decoded codec2 to wav audio
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const wavAudio = await Codec2Lib.rawToWav(decoded);
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// play wav audio buffer
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let audioCtx = new AudioContext()
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const audioBuffer = await audioCtx.decodeAudioData(wavAudio.buffer);
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const sampleSource = audioCtx.createBufferSource();
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sampleSource.buffer = audioBuffer;
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sampleSource.connect(audioCtx.destination)
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sampleSource.start(0);
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};
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// record mic to send to websocket
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await startRecordingMicrophone((encoded) => {
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// do nothing if websocket closed
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if(callWebsocket.readyState !== WebSocket.OPEN){
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return;
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}
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// do nothing when audio muted
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if(checkboxMuteMicElement.checked){
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return;
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}
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// send encoded audio to websocket
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callWebsocket.send(encoded);
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// update stats
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encodedBytesSent += encoded.length;
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encodedBytesSentElement.innerText = formatBytes(encodedBytesSent);
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});
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} catch(error) {
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alert(error);
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console.log(error);
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}
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}
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function stopStreaming() {
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// disconnect websocket
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if(callWebsocket){
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callWebsocket.close()
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}
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// disconnect media stream source
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if(mediaStreamSource){
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mediaStreamSource.disconnect();
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}
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// stop using microphone
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if(microphoneMediaStream){
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microphoneMediaStream.getTracks().forEach(track => track.stop());
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}
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// disconnect the audio worklet node
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if(audioWorkletNode){
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audioWorkletNode.disconnect();
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}
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// close audio context
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if(audioContext){
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audioContext.close();
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}
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}
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async function startListening() {
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// connect to websocket to get codec2 packets
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listenWebsocket = new WebSocket(location.origin.replace(/^http/, 'ws') + "/call/listen");
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listenWebsocket.onmessage = async function(event) {
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// get encoded codec2 bytes from websocket message
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const encoded = await event.data.arrayBuffer();
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|
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// decode codec2 audio
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const decoded = await Codec2Lib.runDecode(codecModeElement.value, encoded);
|
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|
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// convert decoded codec2 to wav audio
|
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const wavAudio = await Codec2Lib.rawToWav(decoded);
|
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|
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// play wav audio buffer
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let audioCtx = new AudioContext()
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const audioBuffer = await audioCtx.decodeAudioData(wavAudio.buffer);
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const sampleSource = audioCtx.createBufferSource();
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sampleSource.buffer = audioBuffer;
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sampleSource.connect(audioCtx.destination)
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sampleSource.start(0);
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|
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};
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|
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// record mic to send to websocket
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await startRecordingMicrophone((encoded) => {
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// do nothing if websocket closed
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if(listenWebsocket.readyState !== WebSocket.OPEN){
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return;
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}
|
||||
|
||||
// do nothing when audio muted
|
||||
if(checkboxMuteMicElement.checked){
|
||||
return;
|
||||
}
|
||||
|
||||
// send encoded audio to websocket
|
||||
listenWebsocket.send(encoded);
|
||||
|
||||
// update stats
|
||||
encodedBytesSent += encoded.length;
|
||||
encodedBytesSentElement.innerText = formatBytes(encodedBytesSent);
|
||||
|
||||
});
|
||||
|
||||
}
|
||||
|
||||
function stopListening() {
|
||||
if(listenWebsocket){
|
||||
listenWebsocket.close();
|
||||
listenWebsocket = null;
|
||||
}
|
||||
}
|
||||
|
||||
function encodeWAV(samples, sampleRate = 8000, numChannels = 1) {
|
||||
},
|
||||
encodeWAV: function(samples, sampleRate = 8000, numChannels = 1) {
|
||||
|
||||
const buffer = new ArrayBuffer(44 + samples.length * 2);
|
||||
const view = new DataView(buffer);
|
||||
|
||||
// RIFF chunk descriptor
|
||||
writeString(view, 0, 'RIFF');
|
||||
this.writeString(view, 0, 'RIFF');
|
||||
view.setUint32(4, 36 + samples.length * 2, true); // file length
|
||||
writeString(view, 8, 'WAVE');
|
||||
this.writeString(view, 8, 'WAVE');
|
||||
|
||||
// fmt sub-chunk
|
||||
writeString(view, 12, 'fmt ');
|
||||
this.writeString(view, 12, 'fmt ');
|
||||
view.setUint32(16, 16, true); // sub-chunk size
|
||||
view.setUint16(20, 1, true); // audio format (1 = PCM)
|
||||
view.setUint16(22, numChannels, true); // number of channels
|
||||
@@ -256,30 +298,27 @@
|
||||
view.setUint16(34, 16, true); // bits per sample
|
||||
|
||||
// data sub-chunk
|
||||
writeString(view, 36, 'data');
|
||||
this.writeString(view, 36, 'data');
|
||||
view.setUint32(40, samples.length * 2, true); // data chunk length
|
||||
|
||||
// write the PCM samples
|
||||
floatTo16BitPCM(view, 44, samples);
|
||||
this.floatTo16BitPCM(view, 44, samples);
|
||||
|
||||
return buffer;
|
||||
|
||||
}
|
||||
|
||||
function writeString(view, offset, string) {
|
||||
},
|
||||
writeString: function(view, offset, string) {
|
||||
for(let i = 0; i < string.length; i++){
|
||||
view.setUint8(offset + i, string.charCodeAt(i));
|
||||
}
|
||||
}
|
||||
|
||||
function floatTo16BitPCM(output, offset, input) {
|
||||
},
|
||||
floatTo16BitPCM: function(output, offset, input) {
|
||||
for(let i = 0; i < input.length; i++, offset += 2){
|
||||
const s = Math.max(-1, Math.min(1, input[i]));
|
||||
output.setInt16(offset, s < 0 ? s * 0x8000 : s * 0x7FFF, true);
|
||||
}
|
||||
}
|
||||
|
||||
function formatBytes(bytes) {
|
||||
},
|
||||
formatBytes: function(bytes) {
|
||||
|
||||
if(bytes === 0){
|
||||
return '0 Bytes';
|
||||
@@ -293,8 +332,9 @@
|
||||
|
||||
return parseFloat((bytes / Math.pow(k, i)).toFixed(decimals)) + ' ' + sizes[i];
|
||||
|
||||
}
|
||||
|
||||
},
|
||||
},
|
||||
}).mount('#app');
|
||||
</script>
|
||||
</body>
|
||||
</html>
|
||||
195
src/audio_call_manager.py
Normal file
195
src/audio_call_manager.py
Normal file
@@ -0,0 +1,195 @@
|
||||
import asyncio
|
||||
from typing import List
|
||||
|
||||
import RNS
|
||||
|
||||
# todo optionally identity self over link
|
||||
# todo allowlist/denylist for incoming calls
|
||||
|
||||
|
||||
class AudioCall:
|
||||
|
||||
def __init__(self, link: RNS.Link, is_outbound: bool):
|
||||
self.link = link
|
||||
self.is_outbound = is_outbound
|
||||
self.link.set_link_closed_callback(self.on_link_closed)
|
||||
self.link.set_packet_callback(self.on_packet)
|
||||
self.audio_packet_listeners = []
|
||||
|
||||
def register_audio_packet_listener(self, callback):
|
||||
self.audio_packet_listeners.append(callback)
|
||||
|
||||
def unregister_audio_packet_listener(self, callback):
|
||||
self.audio_packet_listeners.remove(callback)
|
||||
|
||||
# handle link being closed
|
||||
def on_link_closed(self):
|
||||
print("[AudioCall] on_link_closed")
|
||||
self.hangup()
|
||||
|
||||
# handle packet received over link
|
||||
def on_packet(self, message, packet):
|
||||
|
||||
# send audio received from call initiator to all audio packet listeners
|
||||
for audio_packet_listener in self.audio_packet_listeners:
|
||||
audio_packet_listener(message)
|
||||
|
||||
# send an audio packet over the link
|
||||
def send_audio_packet(self, data):
|
||||
|
||||
# do nothing if link is not active
|
||||
if self.is_active() is False:
|
||||
return
|
||||
|
||||
# drop audio packet if it is too big to send
|
||||
if len(data) > RNS.Link.MDU:
|
||||
print("[AudioCall] dropping audio packet " + str(len(data)) + " bytes exceeds the link packet MDU of " + str(RNS.Link.MDU) + " bytes")
|
||||
return
|
||||
|
||||
# send codec2 audio received from call receiver to call initiator over reticulum link
|
||||
RNS.Packet(self.link, data).send()
|
||||
|
||||
# gets the identity of the caller, or returns None if they did not identify
|
||||
def initiator_identity(self):
|
||||
return self.link.get_remote_identity()
|
||||
|
||||
# determine if this call is still active
|
||||
def is_active(self):
|
||||
return self.link.status == RNS.Link.ACTIVE
|
||||
|
||||
# handle hanging up the call
|
||||
def hangup(self):
|
||||
print("[AudioCall] hangup")
|
||||
self.link.teardown()
|
||||
pass
|
||||
|
||||
|
||||
class AudioCallManager:
|
||||
|
||||
def __init__(self, identity: RNS.Identity):
|
||||
|
||||
self.identity = identity
|
||||
self.on_incoming_call_callback = None
|
||||
self.on_outgoing_call_callback = None
|
||||
self.audio_call_receiver = AudioCallReceiver(manager=self)
|
||||
|
||||
# remember audio calls
|
||||
self.audio_calls: List[AudioCall] = []
|
||||
|
||||
# announces the audio call destination
|
||||
def announce(self, app_data=None):
|
||||
self.audio_call_receiver.destination.announce(app_data)
|
||||
print("[AudioCallManager] announced destination: " + RNS.prettyhexrep(self.audio_call_receiver.destination.hash))
|
||||
|
||||
# set the callback for incoming calls
|
||||
def register_incoming_call_callback(self, callback):
|
||||
self.on_incoming_call_callback = callback
|
||||
|
||||
# set the callback for outgoing calls
|
||||
def register_outgoing_call_callback(self, callback):
|
||||
self.on_outgoing_call_callback = callback
|
||||
|
||||
# handle incoming calls from audio call receiver
|
||||
def handle_incoming_call(self, audio_call: AudioCall):
|
||||
|
||||
# remember it
|
||||
self.audio_calls.append(audio_call)
|
||||
|
||||
# fire callback
|
||||
if self.on_incoming_call_callback is not None:
|
||||
self.on_incoming_call_callback(audio_call)
|
||||
|
||||
# handle outgoing calls
|
||||
def handle_outgoing_call(self, audio_call: AudioCall):
|
||||
|
||||
# remember it
|
||||
self.audio_calls.append(audio_call)
|
||||
|
||||
# fire callback
|
||||
if self.on_outgoing_call_callback is not None:
|
||||
self.on_outgoing_call_callback(audio_call)
|
||||
|
||||
# find an existing audio call from the provided link hash
|
||||
def find_audio_call_by_link_hash(self, link_hash: bytes):
|
||||
for audio_call in self.audio_calls:
|
||||
if audio_call.link.hash == link_hash:
|
||||
return audio_call
|
||||
return None
|
||||
|
||||
# attempts to initiate a call to the provided destination and returns the link hash on success
|
||||
# FIXME: implement timeout. at the moment, it loops forever if no path is found
|
||||
async def initiate(self, destination_hash: bytes) -> bytes:
|
||||
|
||||
# wait until we have a path to the destination
|
||||
# FIXME: implement timeout instead of looping forever
|
||||
if not RNS.Transport.has_path(destination_hash):
|
||||
RNS.Transport.request_path(destination_hash)
|
||||
while not RNS.Transport.has_path(destination_hash):
|
||||
await asyncio.sleep(0.1)
|
||||
|
||||
# create outbound destination to initiate audio calls
|
||||
server_identity = RNS.Identity.recall(destination_hash)
|
||||
server_destination = RNS.Destination(
|
||||
server_identity,
|
||||
RNS.Destination.OUT,
|
||||
RNS.Destination.SINGLE,
|
||||
"call",
|
||||
"audio"
|
||||
)
|
||||
|
||||
# create link
|
||||
link = RNS.Link(server_destination)
|
||||
|
||||
# register link state callbacks
|
||||
link.set_link_established_callback(self.on_link_established)
|
||||
|
||||
return link.hash
|
||||
|
||||
def on_link_established(self, link: RNS.Link):
|
||||
|
||||
# todo: this can be optional, it's only being sent by default for ui, can be removed
|
||||
link.identify(self.identity)
|
||||
|
||||
# create audio call
|
||||
audio_call = AudioCall(link, is_outbound=True)
|
||||
|
||||
# handle new outgoing call
|
||||
self.handle_outgoing_call(audio_call)
|
||||
|
||||
|
||||
class AudioCallReceiver:
|
||||
|
||||
def __init__(self, manager: AudioCallManager):
|
||||
|
||||
self.manager = manager
|
||||
|
||||
# create destination for receiver audio calls
|
||||
self.destination = RNS.Destination(
|
||||
self.manager.identity,
|
||||
RNS.Destination.IN,
|
||||
RNS.Destination.SINGLE,
|
||||
"call",
|
||||
"audio",
|
||||
)
|
||||
|
||||
# register link state callbacks
|
||||
self.destination.set_link_established_callback(self.client_connected)
|
||||
|
||||
# find an existing audio call from the provided link
|
||||
def find_audio_call_by_link_hash(self, link_hash: bytes):
|
||||
for audio_call in self.manager.audio_calls:
|
||||
if audio_call.link.hash == link_hash:
|
||||
return audio_call
|
||||
return None
|
||||
|
||||
# client connected to us, set up an audio call instance
|
||||
def client_connected(self, link: RNS.Link):
|
||||
|
||||
# todo: this can be optional, it's only being sent by default for ui, can be removed
|
||||
link.identify(self.manager.identity)
|
||||
|
||||
# create audio call
|
||||
audio_call = AudioCall(link, is_outbound=False)
|
||||
|
||||
# pass to manager
|
||||
self.manager.handle_incoming_call(audio_call)
|
||||
163
web.py
163
web.py
@@ -18,6 +18,7 @@ from peewee import SqliteDatabase
|
||||
|
||||
import database
|
||||
from lxmf_message_fields import LxmfImageField, LxmfFileAttachmentsField, LxmfFileAttachment
|
||||
from src.audio_call_manager import AudioCall, AudioCallManager
|
||||
|
||||
|
||||
class ReticulumWebChat:
|
||||
@@ -84,7 +85,13 @@ class ReticulumWebChat:
|
||||
# remember websocket clients
|
||||
self.websocket_clients: List[web.WebSocketResponse] = []
|
||||
|
||||
self.link_call_audio = None
|
||||
# register audio call identity
|
||||
self.audio_call_manager = AudioCallManager(identity=self.identity)
|
||||
self.audio_call_manager.register_incoming_call_callback(self.on_incoming_audio_call)
|
||||
|
||||
# handle receiving a new audio call
|
||||
def on_incoming_audio_call(self, audio_call: AudioCall):
|
||||
print("on_incoming_audio_call: {}".format(audio_call.link.hash.hex()))
|
||||
|
||||
# web server has shutdown, likely ctrl+c, but if we don't do the following, the script never exits
|
||||
async def shutdown(self, app):
|
||||
@@ -232,69 +239,89 @@ class ReticulumWebChat:
|
||||
|
||||
return websocket_response
|
||||
|
||||
# handle websocket clients for listening for calls
|
||||
@routes.get("/call/listen")
|
||||
# get calls
|
||||
@routes.get("/api/v1/calls")
|
||||
async def index(request):
|
||||
|
||||
# get audio calls
|
||||
audio_calls = []
|
||||
for audio_call in self.audio_call_manager.audio_calls:
|
||||
|
||||
# get initiator identity hash
|
||||
initiator_identity_hash = None
|
||||
initiator_identity = audio_call.initiator_identity()
|
||||
if initiator_identity is not None:
|
||||
initiator_identity_hash = initiator_identity.hash.hex()
|
||||
|
||||
audio_calls.append({
|
||||
"hash": audio_call.link.hash.hex(),
|
||||
"initiator_identity_hash": initiator_identity_hash,
|
||||
"is_active": audio_call.is_active(),
|
||||
"is_outbound": audio_call.is_outbound,
|
||||
})
|
||||
|
||||
return web.json_response({
|
||||
"audio_calls": audio_calls,
|
||||
})
|
||||
|
||||
# initiate a call to the provided destination
|
||||
@routes.get("/api/v1/calls/initiate/{destination_hash}")
|
||||
async def index(request):
|
||||
|
||||
# get path params
|
||||
destination_hash = request.match_info.get("destination_hash", "")
|
||||
|
||||
# convert destination hash to bytes
|
||||
destination_hash = bytes.fromhex(destination_hash)
|
||||
|
||||
# initiate audio call
|
||||
link_hash = await self.audio_call_manager.initiate(destination_hash)
|
||||
|
||||
return web.json_response({
|
||||
"hash": link_hash.hex(),
|
||||
})
|
||||
|
||||
# handle websocket client for sending and receiving audio packets in a call
|
||||
@routes.get("/api/v1/calls/{audio_call_link_hash}/audio")
|
||||
async def ws(request):
|
||||
|
||||
# get path params
|
||||
audio_call_link_hash = request.match_info.get("audio_call_link_hash", "")
|
||||
|
||||
# convert hash to bytes
|
||||
audio_call_link_hash = bytes.fromhex(audio_call_link_hash)
|
||||
|
||||
# find audio call
|
||||
audio_call = self.audio_call_manager.find_audio_call_by_link_hash(audio_call_link_hash)
|
||||
if audio_call is None:
|
||||
# fixme: web browser expects websocket, so this won't be useful
|
||||
return web.json_response({
|
||||
"message": "audio call not found",
|
||||
}, status=404)
|
||||
|
||||
# send audio received from call initiator to call receiver websocket
|
||||
def on_audio_packet(data):
|
||||
if websocket_response.closed is False:
|
||||
try:
|
||||
asyncio.run(websocket_response.send_bytes(data))
|
||||
except:
|
||||
# ignore errors sending audio packets to websocket
|
||||
pass
|
||||
|
||||
# register audio packet listener
|
||||
audio_call.register_audio_packet_listener(on_audio_packet)
|
||||
|
||||
# prepare websocket response
|
||||
websocket_response = web.WebSocketResponse()
|
||||
await websocket_response.prepare(request)
|
||||
|
||||
# create destination to allow incoming audio calls
|
||||
server_identity = self.identity
|
||||
server_destination = RNS.Destination(
|
||||
server_identity,
|
||||
RNS.Destination.IN,
|
||||
RNS.Destination.SINGLE,
|
||||
"call",
|
||||
"audio",
|
||||
)
|
||||
|
||||
# client connected to us
|
||||
def client_connected(link):
|
||||
print("client connected")
|
||||
self.link_call_audio = link
|
||||
link.set_link_closed_callback(client_disconnected)
|
||||
link.set_packet_callback(server_packet_received)
|
||||
|
||||
# client disconnected from us
|
||||
def client_disconnected(link):
|
||||
print("client disconnected")
|
||||
self.link_call_audio = None
|
||||
|
||||
# client sent us a packet
|
||||
def server_packet_received(message, packet):
|
||||
|
||||
# send audio received from call initiator to call receiver websocket
|
||||
asyncio.run(websocket_response.send_bytes(message))
|
||||
|
||||
# todo send our audio back to call initiator
|
||||
|
||||
# register link state callbacks
|
||||
server_destination.set_link_established_callback(client_connected)
|
||||
|
||||
# announce our call.audio destination
|
||||
print("call.audio announced and waiting for connection: "+ RNS.prettyhexrep(server_destination.hash))
|
||||
server_destination.announce()
|
||||
|
||||
# handle websocket messages until disconnected
|
||||
# FIXME: we should send a type with the message, so we can send other data as well
|
||||
async for msg in websocket_response:
|
||||
msg: WSMessage = msg
|
||||
if msg.type == WSMsgType.BINARY:
|
||||
try:
|
||||
|
||||
# drop audio packet if it is too big to send
|
||||
if len(msg.data) > RNS.Link.MDU:
|
||||
print("dropping packet " + str(len(msg.data)) + " bytes exceeds the link packet MDU of " + str(RNS.Link.MDU) + " bytes")
|
||||
continue
|
||||
|
||||
# send codec2 audio received from call receiver on websocket, to call initiator over reticulum link
|
||||
if self.link_call_audio is not None:
|
||||
print("sending bytes to call initiator: {}".format(len(msg.data)))
|
||||
RNS.Packet(self.link_call_audio, msg.data).send()
|
||||
else:
|
||||
print("link to call initiator not available")
|
||||
|
||||
audio_call.send_audio_packet(msg.data)
|
||||
except Exception as e:
|
||||
# ignore errors while handling message
|
||||
print("failed to process client message")
|
||||
@@ -303,8 +330,35 @@ class ReticulumWebChat:
|
||||
# ignore errors while handling message
|
||||
print('ws connection error %s' % websocket_response.exception())
|
||||
|
||||
# unregister audio packet handler now that the websocket has been closed
|
||||
audio_call.register_audio_packet_listener(on_audio_packet)
|
||||
|
||||
return websocket_response
|
||||
|
||||
# hangup calls
|
||||
@routes.get("/api/v1/calls/{audio_call_link_hash}/hangup")
|
||||
async def index(request):
|
||||
|
||||
# get path params
|
||||
audio_call_link_hash = request.match_info.get("audio_call_link_hash", "")
|
||||
|
||||
# convert hash to bytes
|
||||
audio_call_link_hash = bytes.fromhex(audio_call_link_hash)
|
||||
|
||||
# find audio call
|
||||
audio_call = self.audio_call_manager.find_audio_call_by_link_hash(audio_call_link_hash)
|
||||
if audio_call is None:
|
||||
return web.json_response({
|
||||
"message": "audio call not found",
|
||||
}, status=404)
|
||||
|
||||
# hangup the call
|
||||
audio_call.hangup()
|
||||
|
||||
return web.json_response({
|
||||
"message": "call has been hungup",
|
||||
})
|
||||
|
||||
# serve announces
|
||||
@routes.get("/api/v1/announces")
|
||||
async def index(request):
|
||||
@@ -575,6 +629,9 @@ class ReticulumWebChat:
|
||||
# send announce for lxmf
|
||||
self.local_lxmf_destination.announce(app_data=self.config.display_name.get().encode("utf-8"))
|
||||
|
||||
# send announce for audio call
|
||||
self.audio_call_manager.announce(app_data=self.config.display_name.get().encode("utf-8"))
|
||||
|
||||
# handle downloading a file from a nomadnet node
|
||||
elif _type == "nomadnet.file.download":
|
||||
|
||||
|
||||
Reference in New Issue
Block a user