Changeset 197020 in webkit

Timestamp:

Feb 24, 2016 12:14:58 AM (8 years ago)

Author:

adam.bergkvist@ericsson.com

Message:

WebRTC: RTCPeerConnection: Sort out responsibilities of close() and stop()
​https://bugs.webkit.org/show_bug.cgi?id=154581

Reviewed by Eric Carlson.

Source/WebCore:

Let RTCPeerConnection::close() contain all teardown logic be called by stop().
close() is also responisble for stopping the PeerConnectionBackend and stopping
all RTCRtpSender objects.

Test coverage:
fast/mediastream/RTCRtpSender-replaceTrack.html (updated)
fast/mediastream/RTCPeerConnection-closed-state.html

• Modules/mediastream/RTCPeerConnection.cpp:

(WebCore::RTCPeerConnection::close):
(WebCore::RTCPeerConnection::stop):
(WebCore::RTCPeerConnection::RTCPeerConnection): Deleted.

• Modules/mediastream/RTCPeerConnection.h:

LayoutTests:

Updated test with replaceTrack() call after the RTCPeerConnection object, that
created the RTCRtpSender, is closed.

• fast/mediastream/RTCRtpSender-replaceTrack-expected.txt:
• fast/mediastream/RTCRtpSender-replaceTrack.html:

Location:

trunk

Files:

• LayoutTests/ChangeLog (modified) (1 diff)
• LayoutTests/fast/mediastream/RTCRtpSender-replaceTrack-expected.txt (modified) (1 diff)
• LayoutTests/fast/mediastream/RTCRtpSender-replaceTrack.html (modified) (2 diffs)
• Source/WebCore/ChangeLog (modified) (1 diff)
• Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp (modified) (2 diffs)
• Source/WebCore/Modules/mediastream/RTCPeerConnection.h (modified) (1 diff)

trunk/LayoutTests/ChangeLog

@@ +1 @@
+2016-02-24  Adam Bergkvist  <adam.bergkvist@ericsson.com>
+
+        WebRTC: RTCPeerConnection: Sort out responsibilities of close() and stop()
+        https://bugs.webkit.org/show_bug.cgi?id=154581
+
+        Reviewed by Eric Carlson.
+
+        Updated test with replaceTrack() call after the RTCPeerConnection object, that
+        created the RTCRtpSender, is closed.
+
+        * fast/mediastream/RTCRtpSender-replaceTrack-expected.txt:
+        * fast/mediastream/RTCRtpSender-replaceTrack.html:
+
 2016-02-23  Chris Dumez  <cdumez@apple.com>
 

trunk/LayoutTests/fast/mediastream/RTCRtpSender-replaceTrack-expected.txt

@@ -17 +17 @@
 Stop sender, and try replacing the track
 PASS promise sender.replaceTrack(audioTrack2) rejected with [object DOMError]
+Create a new sender
+PASS sender = pc.addTrack(audioTrack2, stream) did not throw exception.
+Close pc and try replacing the track
+PASS promise sender.replaceTrack(audioTrack3) rejected with [object DOMError]
 End of promise chain
 PASS successfullyParsed is true

trunk/LayoutTests/fast/mediastream/RTCRtpSender-replaceTrack.html

@@ -30 +30 @@
 
                 audioTrack2 = audioTrack.clone();
+                audioTrack3 = audioTrack.clone();
 
                 shouldBe("pc.getSenders().length", "0");
@@ -56 +57 @@
                     pc.removeTrack(sender);
                     return promiseShouldReject("sender.replaceTrack(audioTrack2)");
+                })
+                .catch(function () {
+                    debug("Create a new sender");
+                    shouldNotThrow("sender = pc.addTrack(audioTrack2, stream)");
+                    debug("Close pc and try replacing the track");
+                    pc.close();
+                    return promiseShouldReject("sender.replaceTrack(audioTrack3)");
                 })
                 .catch(function () {

trunk/Source/WebCore/ChangeLog

@@ +1 @@
+2016-02-24  Adam Bergkvist  <adam.bergkvist@ericsson.com>
+
+        WebRTC: RTCPeerConnection: Sort out responsibilities of close() and stop()
+        https://bugs.webkit.org/show_bug.cgi?id=154581
+
+        Reviewed by Eric Carlson.
+
+        Let RTCPeerConnection::close() contain all teardown logic be called by stop().
+        close() is also responisble for stopping the PeerConnectionBackend and stopping
+        all RTCRtpSender objects.
+
+        Test coverage:
+        fast/mediastream/RTCRtpSender-replaceTrack.html (updated)
+        fast/mediastream/RTCPeerConnection-closed-state.html
+
+        * Modules/mediastream/RTCPeerConnection.cpp:
+        (WebCore::RTCPeerConnection::close):
+        (WebCore::RTCPeerConnection::stop):
+        (WebCore::RTCPeerConnection::RTCPeerConnection): Deleted.
+        * Modules/mediastream/RTCPeerConnection.h:
+
 2016-02-24  Adam Bergkvist  <adam.bergkvist@ericsson.com>
 

trunk/Source/WebCore/Modules/mediastream/RTCPeerConnection.cpp

@@ -80 +80 @@
     , m_iceConnectionState(IceConnectionState::New)
     , m_configuration(WTFMove(configuration))
-    , m_stopped(false)
 {
     Document& document = downcast<Document>(context);
@@ -368 +367 @@
         return;
 
-    m_signalingState = SignalingState::Closed;
-}
-
-void RTCPeerConnection::stop()
-{
-    if (m_stopped)
-        return;
-
-    m_stopped = true;
+    m_backend->stop();
+
     m_iceConnectionState = IceConnectionState::Closed;
     m_signalingState = SignalingState::Closed;
+
+    for (auto& sender : m_senderSet)
+        sender->stop();
+}
+
+void RTCPeerConnection::stop()
+{
+    close();
 }
 

trunk/Source/WebCore/Modules/mediastream/RTCPeerConnection.h

@@ -148 +148 @@
 
     RefPtr<RTCConfiguration> m_configuration;
-
-    bool m_stopped;
 };