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SquashSRS4: Refine SDK
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winlin
parent
a7ab78a588
commit
e50582f9c7
6 changed files with 102 additions and 89 deletions
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@ -29,6 +29,14 @@
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function SrsRtcPublisherAsync() {
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var self = {};
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// https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia
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self.constraints = {
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audio: true,
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video: {
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width: {ideal: 320, max: 576}
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}
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};
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// @see https://github.com/rtcdn/rtcdn-draft
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// @url The WebRTC url to play with, for example:
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// webrtc://r.ossrs.net/live/livestream
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@ -56,9 +64,8 @@ function SrsRtcPublisherAsync() {
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self.pc.addTransceiver("audio", {direction: "sendonly"});
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self.pc.addTransceiver("video", {direction: "sendonly"});
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var stream = await navigator.mediaDevices.getUserMedia(
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{audio: true, video: {width: {max: 320}}}
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);
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var stream = await navigator.mediaDevices.getUserMedia(self.constraints);
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// @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
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stream.getTracks().forEach(function (track) {
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self.pc.addTrack(track);
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@ -500,7 +507,8 @@ function SrsRtcPlayerAsync() {
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function SrsRtcFormatSenders(senders, kind) {
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var codecs = [];
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senders.forEach(function (sender) {
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sender.getParameters().codecs.forEach(function(c) {
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var params = sender.getParameters();
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params && params.codecs && params.codecs.forEach(function(c) {
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if (kind && sender.track.kind !== kind) {
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return;
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}
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@ -64,51 +64,51 @@
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</footer>
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</div>
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<script type="text/javascript">
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$(function(){
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var sdk = null; // Global handler to do cleanup when replaying.
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var startPlay = function() {
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$('#rtc_media_player').show();
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$(function(){
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var sdk = null; // Global handler to do cleanup when replaying.
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var startPlay = function() {
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$('#rtc_media_player').show();
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// Close PC when user replay.
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if (sdk) {
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sdk.close();
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}
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sdk = new SrsRtcPlayerAsync();
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// https://webrtc.org/getting-started/remote-streams
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$('#rtc_media_player').prop('srcObject', sdk.stream);
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// Optional callback, SDK will add track to stream.
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// sdk.ontrack = function (event) { console.log('Got track', event); sdk.stream.addTrack(event.track); };
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// For example: webrtc://r.ossrs.net/live/livestream
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var url = $("#txt_url").val();
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sdk.play(url).then(function(session){
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$('#sessionid').html(session.sessionid);
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$('#simulator-drop').attr('href', session.simulator + '?drop=1&username=' + session.sessionid);
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}).catch(function (reason) {
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sdk.close();
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$('#rtc_media_player').hide();
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console.error(reason);
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});
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};
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$('#rtc_media_player').hide();
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var query = parse_query_string();
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srs_init_rtc("#txt_url", query);
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$("#btn_play").click(function() {
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$('#rtc_media_player').prop('muted', false);
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startPlay();
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});
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if (query.autostart === 'true') {
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$('#rtc_media_player').prop('muted', true);
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console.warn('For autostart, we should mute it, see https://www.jianshu.com/p/c3c6944eed5a ' +
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'or https://developers.google.com/web/updates/2017/09/autoplay-policy-changes#audiovideo_elements');
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startPlay();
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// Close PC when user replay.
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if (sdk) {
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sdk.close();
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}
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sdk = new SrsRtcPlayerAsync();
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// https://webrtc.org/getting-started/remote-streams
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$('#rtc_media_player').prop('srcObject', sdk.stream);
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// Optional callback, SDK will add track to stream.
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// sdk.ontrack = function (event) { console.log('Got track', event); sdk.stream.addTrack(event.track); };
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// For example: webrtc://r.ossrs.net/live/livestream
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var url = $("#txt_url").val();
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sdk.play(url).then(function(session){
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$('#sessionid').html(session.sessionid);
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$('#simulator-drop').attr('href', session.simulator + '?drop=1&username=' + session.sessionid);
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}).catch(function (reason) {
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sdk.close();
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$('#rtc_media_player').hide();
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console.error(reason);
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});
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};
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$('#rtc_media_player').hide();
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var query = parse_query_string();
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srs_init_rtc("#txt_url", query);
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$("#btn_play").click(function() {
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$('#rtc_media_player').prop('muted', false);
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startPlay();
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});
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if (query.autostart === 'true') {
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$('#rtc_media_player').prop('muted', true);
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console.warn('For autostart, we should mute it, see https://www.jianshu.com/p/c3c6944eed5a ' +
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'or https://developers.google.com/web/updates/2017/09/autoplay-policy-changes#audiovideo_elements');
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startPlay();
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}
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});
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</script>
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</body>
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</html>
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@ -68,53 +68,52 @@
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</footer>
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</div>
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<script type="text/javascript">
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var pc = null; // Global handler to do cleanup when replaying.
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$(function(){
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var sdk = null; // Global handler to do cleanup when republishing.
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var startPublish = function() {
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$('#rtc_media_player').show();
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$(function(){
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var sdk = null; // Global handler to do cleanup when republishing.
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var startPublish = function() {
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$('#rtc_media_player').show();
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// Close PC when user replay.
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if (sdk) {
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sdk.close();
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// Close PC when user replay.
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if (sdk) {
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sdk.close();
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}
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sdk = new SrsRtcPublisherAsync();
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// User should set the stream when publish is done, @see https://webrtc.org/getting-started/media-devices
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// However SRS SDK provides a consist API like https://webrtc.org/getting-started/remote-streams
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$('#rtc_media_player').prop('srcObject', sdk.stream);
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// Optional callback, SDK will add track to stream.
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// sdk.ontrack = function (event) { console.log('Got track', event); sdk.stream.addTrack(event.track); };
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// https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs#getting_the_supported_codecs
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sdk.pc.onicegatheringstatechange = function (event) {
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if (sdk.pc.iceGatheringState === "complete") {
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$('#acodecs').html(SrsRtcFormatSenders(sdk.pc.getSenders(), "audio"));
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$('#vcodecs').html(SrsRtcFormatSenders(sdk.pc.getSenders(), "video"));
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}
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sdk = new SrsRtcPublisherAsync();
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// User should set the stream when publish is done, @see https://webrtc.org/getting-started/media-devices
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// However SRS SDK provides a consist API like https://webrtc.org/getting-started/remote-streams
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$('#rtc_media_player').prop('srcObject', sdk.stream);
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// Optional callback, SDK will add track to stream.
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// sdk.ontrack = function (event) { console.log('Got track', event); sdk.stream.addTrack(event.track); };
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// https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs#getting_the_supported_codecs
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sdk.pc.onicegatheringstatechange = function (event) {
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if (sdk.pc.iceGatheringState === "complete") {
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$('#acodecs').html(SrsRtcFormatSenders(sdk.pc.getSenders(), "audio"));
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$('#vcodecs').html(SrsRtcFormatSenders(sdk.pc.getSenders(), "video"));
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}
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};
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// For example: webrtc://r.ossrs.net/live/livestream
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var url = $("#txt_url").val();
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sdk.publish(url).then(function(session){
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$('#sessionid').html(session.sessionid);
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$('#simulator-drop').attr('href', session.simulator + '?drop=1&username=' + session.sessionid);
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}).catch(function (reason) {
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sdk.close();
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$('#rtc_media_player').hide();
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console.error(reason);
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});
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};
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$('#rtc_media_player').hide();
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var query = parse_query_string();
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srs_init_rtc("#txt_url", query);
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// For example: webrtc://r.ossrs.net/live/livestream
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var url = $("#txt_url").val();
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sdk.publish(url).then(function(session){
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$('#sessionid').html(session.sessionid);
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$('#simulator-drop').attr('href', session.simulator + '?drop=1&username=' + session.sessionid);
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}).catch(function (reason) {
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sdk.close();
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$('#rtc_media_player').hide();
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console.error(reason);
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});
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};
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$("#btn_publish").click(startPublish);
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if (query.autostart === 'true') {
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startPublish();
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}
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});
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$('#rtc_media_player').hide();
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var query = parse_query_string();
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srs_init_rtc("#txt_url", query);
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$("#btn_publish").click(startPublish);
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if (query.autostart === 'true') {
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startPublish();
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}
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});
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</script>
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</body>
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</html>
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