325887e325
This is a rework of the audio join procedure whithout the explict listen only separation in mind. It's supposed to be used in conjunction with the transparent listen only feature so that the distinction between modes is seamless with minimal server-side impact. An abridged list of changes: - Let the user pick no input device when joining microphone while allowing them to set an input device on the fly later on - Give the user the option to join audio with no input device whenever we fail to obtain input devices, with the option to try re-enabling them on the fly later on - Add the option to open the audio settings modal (echo test et al) via the in-call device selection chevron - Rework the SFU audio bridge and its services to support adding/removing tracks on the fly without renegotiation - Rework the SFU audio bridge and its services to support a new peer role called "passive-sendrecv". That role is used by dupled peers that have no active input source on start, but might have one later on. - Remove stale PermissionsOverlay component from the audio modal - Rework how permission errors are detected using the Permissions API - Rework the local echo test so that it uses a separate media tag rather than the remote - Add new, separate dialplans that mute/hold FreeSWITCH channels on hold based on UA strings. This is orchestrated server-side via webrtc-sfu and akka-apps. The basic difference here is that channels now join in their desired state rather than waiting for client side observers to sync the state up. It also mitigates transparent listen only performance edge cases on multiple audio channels joining at the same time. The old, decoupled listen only mode is still present in code while we validate this new approach. To test this, transparentListenOnly must be enabled and listen only mode must be disable on audio join so that the user skips straight through microphone join.
134 lines
3.9 KiB
JavaScript
134 lines
3.9 KiB
JavaScript
import LocalPCLoopback from '/imports/ui/services/webrtc-base/local-pc-loopback';
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import browserInfo from '/imports/utils/browserInfo';
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const LOCAL_MEDIA_TAG = '#local-media';
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let audioContext = null;
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let sourceContext = null;
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let contextDestination = null;
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let stubAudioElement = null;
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let delayNode = null;
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const shouldUseRTCLoopback = () => {
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const USE_RTC_LOOPBACK_CHR = window.meetingClientSettings.public.media.localEchoTest.useRtcLoopbackInChromium;
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return (browserInfo.isChrome || browserInfo.isEdge) && USE_RTC_LOOPBACK_CHR;
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};
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const createAudioRTCLoopback = () => new LocalPCLoopback({ audio: true });
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const cleanupDelayNode = () => {
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if (delayNode) {
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delayNode.disconnect();
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delayNode = null;
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}
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if (sourceContext) {
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sourceContext.disconnect();
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sourceContext = null;
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}
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if (audioContext) {
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audioContext.close();
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audioContext = null;
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}
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if (contextDestination) {
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contextDestination.disconnect();
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contextDestination = null;
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}
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if (stubAudioElement) {
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stubAudioElement.pause();
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stubAudioElement.srcObject = null;
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stubAudioElement = null;
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}
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};
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const addDelayNode = (stream) => {
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const {
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delayTime = 0.5,
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maxDelayTime = 2,
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} = window.meetingClientSettings.public.media.localEchoTest.delay;
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if (stream) {
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if (delayNode || audioContext || sourceContext) cleanupDelayNode();
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const audioElement = document.querySelector(LOCAL_MEDIA_TAG);
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// Workaround: attach the stream to a muted stub audio element to be able to play it in
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// Chromium-based browsers. See https://bugs.chromium.org/p/chromium/issues/detail?id=933677
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stubAudioElement = new Audio();
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stubAudioElement.muted = true;
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stubAudioElement.srcObject = stream;
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// Create a new AudioContext to be able to add a delay to the stream
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audioContext = new AudioContext();
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sourceContext = audioContext.createMediaStreamSource(stream);
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contextDestination = audioContext.createMediaStreamDestination();
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// Create a DelayNode to add a delay to the stream
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delayNode = new DelayNode(audioContext, { delayTime, maxDelayTime });
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// Connect the stream to the DelayNode and then to the MediaStreamDestinationNode
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// to be able to play the stream.
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sourceContext.connect(delayNode);
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delayNode.connect(contextDestination);
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delayNode.delayTime.setValueAtTime(delayTime, audioContext.currentTime);
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// Play the stream with the delay in the default audio element (local-media)
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audioElement.srcObject = contextDestination.stream;
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}
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};
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const deattachEchoStream = () => {
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const {
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enabled: DELAY_ENABLED = true,
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} = window.meetingClientSettings.public.media.localEchoTest.delay;
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const audioElement = document.querySelector(LOCAL_MEDIA_TAG);
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if (DELAY_ENABLED) {
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audioElement.muted = false;
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cleanupDelayNode();
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}
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audioElement.pause();
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audioElement.srcObject = null;
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};
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const playEchoStream = async (stream, loopbackAgent = null) => {
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const {
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enabled: DELAY_ENABLED = true,
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} = window.meetingClientSettings.public.media.localEchoTest.delay;
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if (stream) {
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deattachEchoStream();
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let streamToPlay = stream;
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if (loopbackAgent) {
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// Chromium based browsers need audio to go through PCs for echo cancellation
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// to work. See https://bugs.chromium.org/p/chromium/issues/detail?id=687574
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try {
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await loopbackAgent.start(stream);
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streamToPlay = loopbackAgent.loopbackStream;
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} catch (error) {
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loopbackAgent.stop();
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}
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}
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if (DELAY_ENABLED) {
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addDelayNode(streamToPlay);
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} else {
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// No delay: play the stream in the default audio element (local-media),
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// no strings attached.
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const audioElement = document.querySelector(LOCAL_MEDIA_TAG);
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audioElement.srcObject = streamToPlay;
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audioElement.muted = false;
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audioElement.play();
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}
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}
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};
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export default {
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shouldUseRTCLoopback,
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createAudioRTCLoopback,
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deattachEchoStream,
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playEchoStream,
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};
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