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SquashSRS4: Refine SDK

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pull/2373/head
winlin 4 years ago
parent a7ab78a588
commit e50582f9c7
6 changed files with 97 additions and 84 deletions
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2
README.md
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@ -181,6 +181,7 @@ The ports used by SRS:
## V4 changes
* v4.0, 2021-05-21, Fix [#2370][bug #2370] bug for Firefox play stream(published by Chrome). 4.0.121
* v4.0, 2021-05-21, RTC: Refine sdk, migrate from onaddstream to ontrack. 4.0.120
* v4.0, 2021-05-21, Tools: Refine configure options. 4.0.119
* v4.0, 2021-05-20, Fix build fail when disable RTC by --rtc=off. 4.0.118
@ -1924,6 +1925,7 @@ Winlin
[bug #2355]: https://github.com/ossrs/srs/issues/2355
[bug #307]: https://github.com/ossrs/srs/issues/307
[bug #2362]: https://github.com/ossrs/srs/issues/2362
[bug #2370]: https://github.com/ossrs/srs/issues/2370
[bug #yyyyyyyyyyyyy]: https://github.com/ossrs/srs/issues/yyyyyyyyyyyyy
[bug #1631]: https://github.com/ossrs/srs/issues/1631

16
trunk/research/players/js/srs.sdk.js
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@ -29,6 +29,14 @@
function SrsRtcPublisherAsync() {
var self = {};
// https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia
self.constraints = {
audio: true,
video: {
width: {ideal: 320, max: 576}
}
};
// @see https://github.com/rtcdn/rtcdn-draft
// @url The WebRTC url to play with, for example:
// webrtc://r.ossrs.net/live/livestream
@ -56,9 +64,8 @@ function SrsRtcPublisherAsync() {
self.pc.addTransceiver("audio", {direction: "sendonly"});
self.pc.addTransceiver("video", {direction: "sendonly"});
var stream = await navigator.mediaDevices.getUserMedia(
{audio: true, video: {width: {max: 320}}}
);
var stream = await navigator.mediaDevices.getUserMedia(self.constraints);
// @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
stream.getTracks().forEach(function (track) {
self.pc.addTrack(track);
@ -500,7 +507,8 @@ function SrsRtcPlayerAsync() {
function SrsRtcFormatSenders(senders, kind) {
var codecs = [];
senders.forEach(function (sender) {
sender.getParameters().codecs.forEach(function(c) {
var params = sender.getParameters();
params && params.codecs && params.codecs.forEach(function(c) {
if (kind && sender.track.kind !== kind) {
return;
}

76
trunk/research/players/rtc_player.html
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@ -64,51 +64,51 @@
</footer>
</div>
<script type="text/javascript">
$(function(){
var sdk = null; // Global handler to do cleanup when replaying.
var startPlay = function() {
$('#rtc_media_player').show();
$(function(){
var sdk = null; // Global handler to do cleanup when replaying.
var startPlay = function() {
$('#rtc_media_player').show();
// Close PC when user replay.
if (sdk) {
sdk.close();
}
sdk = new SrsRtcPlayerAsync();
// https://webrtc.org/getting-started/remote-streams
$('#rtc_media_player').prop('srcObject', sdk.stream);
// Optional callback, SDK will add track to stream.
// sdk.ontrack = function (event) { console.log('Got track', event); sdk.stream.addTrack(event.track); };
// For example: webrtc://r.ossrs.net/live/livestream
var url = $("#txt_url").val();
sdk.play(url).then(function(session){
$('#sessionid').html(session.sessionid);
$('#simulator-drop').attr('href', session.simulator + '?drop=1&username=' + session.sessionid);
}).catch(function (reason) {
sdk.close();
$('#rtc_media_player').hide();
console.error(reason);
});
};
// Close PC when user replay.
if (sdk) {
sdk.close();
}
sdk = new SrsRtcPlayerAsync();
$('#rtc_media_player').hide();
var query = parse_query_string();
srs_init_rtc("#txt_url", query);
// https://webrtc.org/getting-started/remote-streams
$('#rtc_media_player').prop('srcObject', sdk.stream);
// Optional callback, SDK will add track to stream.
// sdk.ontrack = function (event) { console.log('Got track', event); sdk.stream.addTrack(event.track); };
$("#btn_play").click(function() {
$('#rtc_media_player').prop('muted', false);
startPlay();
// For example: webrtc://r.ossrs.net/live/livestream
var url = $("#txt_url").val();
sdk.play(url).then(function(session){
$('#sessionid').html(session.sessionid);
$('#simulator-drop').attr('href', session.simulator + '?drop=1&username=' + session.sessionid);
}).catch(function (reason) {
sdk.close();
$('#rtc_media_player').hide();
console.error(reason);
});
};
if (query.autostart === 'true') {
$('#rtc_media_player').prop('muted', true);
console.warn('For autostart, we should mute it, see https://www.jianshu.com/p/c3c6944eed5a ' +
'or https://developers.google.com/web/updates/2017/09/autoplay-policy-changes#audiovideo_elements');
$('#rtc_media_player').hide();
var query = parse_query_string();
srs_init_rtc("#txt_url", query);
startPlay();
}
$("#btn_play").click(function() {
$('#rtc_media_player').prop('muted', false);
startPlay();
});
if (query.autostart === 'true') {
$('#rtc_media_player').prop('muted', true);
console.warn('For autostart, we should mute it, see https://www.jianshu.com/p/c3c6944eed5a ' +
'or https://developers.google.com/web/updates/2017/09/autoplay-policy-changes#audiovideo_elements');
startPlay();
}
});
</script>
</body>
</html>

81
trunk/research/players/rtc_publisher.html
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@ -68,53 +68,52 @@
</footer>
</div>
<script type="text/javascript">
var pc = null; // Global handler to do cleanup when replaying.
$(function(){
var sdk = null; // Global handler to do cleanup when republishing.
var startPublish = function() {
$('#rtc_media_player').show();
$(function(){
var sdk = null; // Global handler to do cleanup when republishing.
var startPublish = function() {
$('#rtc_media_player').show();
// Close PC when user replay.
if (sdk) {
sdk.close();
}
sdk = new SrsRtcPublisherAsync();
// User should set the stream when publish is done, @see https://webrtc.org/getting-started/media-devices
// However SRS SDK provides a consist API like https://webrtc.org/getting-started/remote-streams
$('#rtc_media_player').prop('srcObject', sdk.stream);
// Optional callback, SDK will add track to stream.
// sdk.ontrack = function (event) { console.log('Got track', event); sdk.stream.addTrack(event.track); };
// Close PC when user replay.
if (sdk) {
sdk.close();
}
sdk = new SrsRtcPublisherAsync();
// https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs#getting_the_supported_codecs
sdk.pc.onicegatheringstatechange = function (event) {
if (sdk.pc.iceGatheringState === "complete") {
$('#acodecs').html(SrsRtcFormatSenders(sdk.pc.getSenders(), "audio"));
$('#vcodecs').html(SrsRtcFormatSenders(sdk.pc.getSenders(), "video"));
}
};
// User should set the stream when publish is done, @see https://webrtc.org/getting-started/media-devices
// However SRS SDK provides a consist API like https://webrtc.org/getting-started/remote-streams
$('#rtc_media_player').prop('srcObject', sdk.stream);
// Optional callback, SDK will add track to stream.
// sdk.ontrack = function (event) { console.log('Got track', event); sdk.stream.addTrack(event.track); };
// For example: webrtc://r.ossrs.net/live/livestream
var url = $("#txt_url").val();
sdk.publish(url).then(function(session){
$('#sessionid').html(session.sessionid);
$('#simulator-drop').attr('href', session.simulator + '?drop=1&username=' + session.sessionid);
}).catch(function (reason) {
sdk.close();
$('#rtc_media_player').hide();
console.error(reason);
});
// https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs#getting_the_supported_codecs
sdk.pc.onicegatheringstatechange = function (event) {
if (sdk.pc.iceGatheringState === "complete") {
$('#acodecs').html(SrsRtcFormatSenders(sdk.pc.getSenders(), "audio"));
$('#vcodecs').html(SrsRtcFormatSenders(sdk.pc.getSenders(), "video"));
}
};
$('#rtc_media_player').hide();
var query = parse_query_string();
srs_init_rtc("#txt_url", query);
// For example: webrtc://r.ossrs.net/live/livestream
var url = $("#txt_url").val();
sdk.publish(url).then(function(session){
$('#sessionid').html(session.sessionid);
$('#simulator-drop').attr('href', session.simulator + '?drop=1&username=' + session.sessionid);
}).catch(function (reason) {
sdk.close();
$('#rtc_media_player').hide();
console.error(reason);
});
};
$("#btn_publish").click(startPublish);
if (query.autostart === 'true') {
startPublish();
}
});
$('#rtc_media_player').hide();
var query = parse_query_string();
srs_init_rtc("#txt_url", query);
$("#btn_publish").click(startPublish);
if (query.autostart === 'true') {
startPublish();
}
});
</script>
</body>
</html>

4
trunk/src/app/srs_app_rtc_conn.cpp
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@ -3096,6 +3096,10 @@ srs_error_t SrsRtcConnection::negotiate_play_capability(SrsRtcUserConfig* ruc, s
for (int j = 0; j < (int)track_descs.size(); ++j) {
SrsRtcTrackDescription* track = track_descs.at(j)->copy();
// We should clear the extmaps of source(publisher).
// @see https://github.com/ossrs/srs/issues/2370
track->extmaps_.clear();
// Use remote/source/offer PayloadType.
track->media_->pt_of_publisher_ = track->media_->pt_;
track->media_->pt_ = remote_payload.payload_type_;

2
trunk/src/core/srs_core_version4.hpp
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@ -26,6 +26,6 @@
#define VERSION_MAJOR 4
#define VERSION_MINOR 0
#define VERSION_REVISION 120
#define VERSION_REVISION 121
#endif

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