[GStreamer][WebRTC] Deadlocks web process at shutdown

https://bugs.webkit.org/show_bug.cgi?id=259973 Reviewed by Carlos Garcia Campos. When stopping outgoing media sources we were sending an EOS event to the mediastreamsrc, but not to the fallback source, so the downstream input-selector was blocked forever, waiting the EOS event from that source. * Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceGStreamer.cpp: (WebCore::RealtimeOutgoingAudioSourceGStreamer::stopFallbackSource): * Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceGStreamer.h: * Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingMediaSourceGStreamer.cpp: (WebCore::RealtimeOutgoingMediaSourceGStreamer::stopOutgoingSource): * Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingMediaSourceGStreamer.h: (WebCore::RealtimeOutgoingMediaSourceGStreamer::stopFallbackSource): * Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingVideoSourceGStreamer.cpp: (WebCore::RealtimeOutgoingVideoSourceGStreamer::stopFallbackSource): * Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingVideoSourceGStreamer.h: Canonical link: https://commits.webkit.org/266759@main
WebKit · Aug 10, 2023 · e3416ac · e3416ac
1 parent a36c08c
commit e3416ac
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Showing 4 changed files with 11 additions and 2 deletions.
diff --git a/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceGStreamer.h b/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceGStreamer.h
@@ -43,7 +43,6 @@ class RealtimeOutgoingAudioSourceGStreamer final : public RealtimeOutgoingMediaS
     void unlinkOutgoingSource() final;
     void linkOutgoingSource() final;
 
-    GRefPtr<GstElement> m_fallbackSource;
     GRefPtr<GstElement> m_audioconvert;
     GRefPtr<GstElement> m_audioresample;
 };

diff --git a/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingMediaSourceGStreamer.cpp b/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingMediaSourceGStreamer.cpp
@@ -50,6 +50,7 @@ RealtimeOutgoingMediaSourceGStreamer::RealtimeOutgoingMediaSourceGStreamer(const
     m_bin = gst_bin_new(nullptr);
 
     m_inputSelector = gst_element_factory_make("input-selector", nullptr);
+    gst_util_set_object_arg(G_OBJECT(m_inputSelector.get()), "sync-mode", "clock");
 
     m_preEncoderQueue = gst_element_factory_make("queue", nullptr);
     m_postEncoderQueue = gst_element_factory_make("queue", nullptr);
@@ -70,6 +71,15 @@ RealtimeOutgoingMediaSourceGStreamer::~RealtimeOutgoingMediaSourceGStreamer()
     if (m_transceiver)
         g_signal_handlers_disconnect_by_data(m_transceiver.get(), this);
 
+    if (m_fallbackSource) {
+        gst_element_set_locked_state(m_fallbackSource.get(), TRUE);
+        gst_element_set_state(m_fallbackSource.get(), GST_STATE_READY);
+        gst_element_unlink(m_fallbackSource.get(), m_inputSelector.get());
+        gst_element_set_state(m_fallbackSource.get(), GST_STATE_NULL);
+        gst_element_release_request_pad(m_inputSelector.get(), m_fallbackPad.get());
+        gst_element_set_locked_state(m_fallbackSource.get(), FALSE);
+    }
+
     stopOutgoingSource();
 
     if (GST_IS_PAD(m_webrtcSinkPad.get())) {

diff --git a/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingMediaSourceGStreamer.h b/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingMediaSourceGStreamer.h
@@ -91,6 +91,7 @@ class RealtimeOutgoingMediaSourceGStreamer : public ThreadSafeRefCounted<Realtim
     GRefPtr<GstPad> m_webrtcSinkPad;
     RefPtr<UniqueSSRCGenerator> m_ssrcGenerator;
     GUniquePtr<GstStructure> m_parameters;
+    GRefPtr<GstElement> m_fallbackSource;
 
 private:
     void sourceMutedChanged();

diff --git a/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingVideoSourceGStreamer.h b/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingVideoSourceGStreamer.h
@@ -61,7 +61,6 @@ class RealtimeOutgoingVideoSourceGStreamer final : public RealtimeOutgoingMediaS
 
     void updateStats(GstBuffer*);
 
-    GRefPtr<GstElement> m_fallbackSource;
     GRefPtr<GstElement> m_videoConvert;
     GRefPtr<GstElement> m_videoFlip;
     GRefPtr<GstElement> m_videoRate;