From e50582f9c72258bfb057cf6a143f9cf1f2b1e54d Mon Sep 17 00:00:00 2001
From: winlin <winlin@vip.126.com>
Date: Fri, 21 May 2021 19:57:59 +0800
Subject: [PATCH] SquashSRS4: Refine SDK

---
 README.md                                 |  2 +
 trunk/research/players/js/srs.sdk.js      | 16 +++--
 trunk/research/players/rtc_player.html    | 84 +++++++++++------------
 trunk/research/players/rtc_publisher.html | 83 +++++++++++-----------
 trunk/src/app/srs_app_rtc_conn.cpp        |  4 ++
 trunk/src/core/srs_core_version4.hpp      |  2 +-
 6 files changed, 102 insertions(+), 89 deletions(-)

diff --git a/README.md b/README.md
index af1f15bd9..de3266765 100755
--- a/README.md
+++ b/README.md
@@ -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
diff --git a/trunk/research/players/js/srs.sdk.js b/trunk/research/players/js/srs.sdk.js
index 860735413..2e3cd87d1 100644
--- a/trunk/research/players/js/srs.sdk.js
+++ b/trunk/research/players/js/srs.sdk.js
@@ -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;
             }
diff --git a/trunk/research/players/rtc_player.html b/trunk/research/players/rtc_player.html
index 0a4c3a3a5..c1e574e23 100644
--- a/trunk/research/players/rtc_player.html
+++ b/trunk/research/players/rtc_player.html
@@ -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);
-            });
-        };
-
-        $('#rtc_media_player').hide();
-        var query = parse_query_string();
-        srs_init_rtc("#txt_url", query);
-
-        $("#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();
+        // 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);
+        });
+    };
+
+    $('#rtc_media_player').hide();
+    var query = parse_query_string();
+    srs_init_rtc("#txt_url", query);
+
+    $("#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>
diff --git a/trunk/research/players/rtc_publisher.html b/trunk/research/players/rtc_publisher.html
index 1159d0e8f..3d1681051 100644
--- a/trunk/research/players/rtc_publisher.html
+++ b/trunk/research/players/rtc_publisher.html
@@ -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();
+        // 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); };
+
+        // 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"));
             }
-            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); };
-
-            // 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"));
-                }
-            };
-
-            // 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);
-            });
         };
 
-        $('#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>
diff --git a/trunk/src/app/srs_app_rtc_conn.cpp b/trunk/src/app/srs_app_rtc_conn.cpp
index a2cac352c..96a1a9714 100644
--- a/trunk/src/app/srs_app_rtc_conn.cpp
+++ b/trunk/src/app/srs_app_rtc_conn.cpp
@@ -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_;
diff --git a/trunk/src/core/srs_core_version4.hpp b/trunk/src/core/srs_core_version4.hpp
index f4e0917d0..44eec8dc1 100644
--- a/trunk/src/core/srs_core_version4.hpp
+++ b/trunk/src/core/srs_core_version4.hpp
@@ -26,6 +26,6 @@
 
 #define VERSION_MAJOR       4
 #define VERSION_MINOR       0
-#define VERSION_REVISION    120
+#define VERSION_REVISION    121
 
 #endif