diff --git a/trunk/doc/CHANGELOG.md b/trunk/doc/CHANGELOG.md
index 18a7b36a4..c831ec94b 100644
--- a/trunk/doc/CHANGELOG.md
+++ b/trunk/doc/CHANGELOG.md
@@ -7,6 +7,7 @@ The changelog for SRS.
## SRS 7.0 Changelog
+* v7.0, 2024-10-15, Merge [#4160](https://github.com/ossrs/srs/pull/4160): RTC2RTMP: Fix screen sharing stutter caused by packet loss. v7.0.18 (#4160)
* v7.0, 2024-10-15, Merge [#3979](https://github.com/ossrs/srs/pull/3979): ST: Use clock_gettime to prevent time jumping backwards. v7.0.17 (#3979)
* v7.0, 2024-09-09, Merge [#4158](https://github.com/ossrs/srs/pull/4158): Proxy: Support proxy server for SRS. v7.0.16 (#4158)
* v7.0, 2024-09-09, Merge [#4171](https://github.com/ossrs/srs/pull/4171): Heartbeat: Report ports for proxy server. v7.0.15 (#4171)
@@ -29,6 +30,7 @@ The changelog for SRS.
## SRS 6.0 Changelog
+* v6.0, 2024-10-15, Merge [#4160](https://github.com/ossrs/srs/pull/4160): RTC2RTMP: Fix screen sharing stutter caused by packet loss. v6.0.157 (#4160)
* v6.0, 2024-09-09, Merge [#4171](https://github.com/ossrs/srs/pull/4171): Heartbeat: Report ports for proxy server. v6.0.156 (#4171)
* v6.0, 2024-09-01, Merge [#4165](https://github.com/ossrs/srs/pull/4165): FLV: Refine source and http handler. v6.0.155 (#4165)
* v6.0, 2024-09-01, Merge [#4166](https://github.com/ossrs/srs/pull/4166): Edge: Fix flv edge crash when http unmount. v6.0.154 (#4166)
@@ -189,6 +191,7 @@ The changelog for SRS.
## SRS 5.0 Changelog
+* v5.0, 2024-10-15, Merge [#4160](https://github.com/ossrs/srs/pull/4160): RTC2RTMP: Fix screen sharing stutter caused by packet loss. v5.0.216 (#4160)
* v5.0, 2024-09-09, Merge [#4171](https://github.com/ossrs/srs/pull/4171): Heartbeat: Report ports for proxy server. v5.0.215 (#4171)
* v5.0, 2024-07-24, Merge [#4126](https://github.com/ossrs/srs/pull/4126): Edge: Improve stability for state and fd closing. v5.0.214 (#4126)
* v5.0, 2024-06-03, Merge [#4057](https://github.com/ossrs/srs/pull/4057): RTC: Support dropping h.264 SEI from NALUs. v5.0.213 (#4057)
diff --git a/trunk/research/players/js/srs.sdk.js b/trunk/research/players/js/srs.sdk.js
index 2a597883a..0e0c4fd47 100644
--- a/trunk/research/players/js/srs.sdk.js
+++ b/trunk/research/players/js/srs.sdk.js
@@ -527,36 +527,56 @@ function SrsRtcWhipWhepAsync() {
// @url The WebRTC url to publish with, for example:
// http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream
// @options The options to control playing, supports:
- // videoOnly: boolean, whether only play video, default to false.
- // audioOnly: boolean, whether only play audio, default to false.
+ // camera: boolean, whether capture video from camera, default to true.
+ // screen: boolean, whether capture video from screen, default to false.
+ // audio: boolean, whether play audio, default to true.
self.publish = async function (url, options) {
if (url.indexOf('/whip/') === -1) throw new Error(`invalid WHIP url ${url}`);
- if (options?.videoOnly && options?.audioOnly) throw new Error(`The videoOnly and audioOnly in options can't be true at the same time`);
+ const hasAudio = options?.audio ?? true;
+ const useCamera = options?.camera ?? true;
+ const useScreen = options?.screen ?? false;
+
+ if (!hasAudio && !useCamera && !useScreen) throw new Error(`The camera, screen and audio can't be false at the same time`);
- if (!options?.videoOnly) {
+ if (hasAudio) {
self.pc.addTransceiver("audio", {direction: "sendonly"});
} else {
self.constraints.audio = false;
}
- if (!options?.audioOnly) {
+ if (useCamera || useScreen) {
self.pc.addTransceiver("video", {direction: "sendonly"});
- } else {
+ }
+
+ if (!useCamera) {
self.constraints.video = false;
}
if (!navigator.mediaDevices && window.location.protocol === 'http:' && window.location.hostname !== 'localhost') {
throw new SrsError('HttpsRequiredError', `Please use HTTPS or localhost to publish, read https://github.com/ossrs/srs/issues/2762#issuecomment-983147576`);
}
- 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);
+ if (useScreen) {
+ const displayStream = await navigator.mediaDevices.getDisplayMedia({
+ video: true
+ });
+ // @see https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/addStream#Migrating_to_addTrack
+ displayStream.getTracks().forEach(function (track) {
+ self.pc.addTrack(track);
+ // Notify about local track when stream is ok.
+ self.ontrack && self.ontrack({track: track});
+ });
+ }
- // Notify about local track when stream is ok.
- self.ontrack && self.ontrack({track: track});
- });
+ if (useCamera || hasAudio) {
+ const userStream = await navigator.mediaDevices.getUserMedia(self.constraints);
+
+ userStream.getTracks().forEach(function (track) {
+ self.pc.addTrack(track);
+ // Notify about local track when stream is ok.
+ self.ontrack && self.ontrack({track: track});
+ });
+ }
var offer = await self.pc.createOffer();
await self.pc.setLocalDescription(offer);
diff --git a/trunk/research/players/whip.html b/trunk/research/players/whip.html
index 60067fe7b..11e4ac993 100644
--- a/trunk/research/players/whip.html
+++ b/trunk/research/players/whip.html
@@ -62,10 +62,16 @@
Controls:
+
+
@@ -113,8 +119,9 @@ $(function(){
// For example: webrtc://r.ossrs.net/live/livestream
var url = $("#txt_url").val();
sdk.publish(url, {
- videoOnly: $('#ch_videoonly').prop('checked'),
- audioOnly: $('#ch_audioonly').prop('checked'),
+ camera: $('#ra_camera').prop('checked'),
+ screen: $('#ra_screen').prop('checked'),
+ audio: $('#ch_audio').prop('checked')
}).then(function(session){
$('#sessionid').html(session.sessionid);
$('#simulator-drop').attr('href', session.simulator + '?drop=1&username=' + session.sessionid);
diff --git a/trunk/src/app/srs_app_rtc_source.cpp b/trunk/src/app/srs_app_rtc_source.cpp
index 8aebf5d53..5ad61e3f9 100644
--- a/trunk/src/app/srs_app_rtc_source.cpp
+++ b/trunk/src/app/srs_app_rtc_source.cpp
@@ -1775,6 +1775,23 @@ srs_error_t SrsRtcFrameBuilder::packet_video_rtmp(const uint16_t start, const ui
if (0 == nb_payload) {
srs_warn("empty nalu");
+
+ // The chrome web browser send RTP packet with empty payload frequently,
+ // reset header_sn_, lost_sn_ and continue to found next frame in this case,
+ // otherwise, all the cached RTP packets are dropped before next key frame arrive.
+ header_sn_ = end + 1;
+ uint16_t tail_sn = 0;
+ int sn = find_next_lost_sn(header_sn_, tail_sn);
+ if (-1 == sn) {
+ if (check_frame_complete(header_sn_, tail_sn)) {
+ err = packet_video_rtmp(header_sn_, tail_sn);
+ }
+ } else if (-2 == sn) {
+ return srs_error_new(ERROR_RTC_RTP_MUXER, "video cache is overflow");
+ } else {
+ lost_sn_ = sn;
+ }
+
return err;
}
diff --git a/trunk/src/core/srs_core_version5.hpp b/trunk/src/core/srs_core_version5.hpp
index 791b3ae6e..1f5323e3f 100644
--- a/trunk/src/core/srs_core_version5.hpp
+++ b/trunk/src/core/srs_core_version5.hpp
@@ -9,6 +9,6 @@
#define VERSION_MAJOR 5
#define VERSION_MINOR 0
-#define VERSION_REVISION 215
+#define VERSION_REVISION 216
#endif
diff --git a/trunk/src/core/srs_core_version6.hpp b/trunk/src/core/srs_core_version6.hpp
index 69725cd53..f7d8dce7b 100644
--- a/trunk/src/core/srs_core_version6.hpp
+++ b/trunk/src/core/srs_core_version6.hpp
@@ -9,6 +9,6 @@
#define VERSION_MAJOR 6
#define VERSION_MINOR 0
-#define VERSION_REVISION 156
+#define VERSION_REVISION 157
#endif
diff --git a/trunk/src/core/srs_core_version7.hpp b/trunk/src/core/srs_core_version7.hpp
index 67461fab9..c4d31842c 100644
--- a/trunk/src/core/srs_core_version7.hpp
+++ b/trunk/src/core/srs_core_version7.hpp
@@ -9,6 +9,6 @@
#define VERSION_MAJOR 7
#define VERSION_MINOR 0
-#define VERSION_REVISION 17
+#define VERSION_REVISION 18
#endif
\ No newline at end of file