[GStreamer][WebRTC] An audio track should be muted when capture is interrupted by the OS.

philn · webkit-commit-queue · commit d1f397a67092 · 2021-04-12T17:18:28.000Z
https://bugs.webkit.org/show_bug.cgi?id=196606 Patch by Philippe Normand <pnormand@igalia.com> on 2021-04-12 Reviewed by Xabier Rodriguez-Calvar. Source/WebCore: Add basic interruption support in the audio capture source and mock audio capture source. * platform/mediastream/gstreamer/GStreamerAudioCaptureSource.cpp: (WebCore::GStreamerAudioCaptureSource::interrupted const): (WebCore::GStreamerAudioCaptureSource::setInterruptedForTesting): * platform/mediastream/gstreamer/GStreamerAudioCaptureSource.h: * platform/mediastream/gstreamer/GStreamerCapturer.cpp: (WebCore::GStreamerCapturer::setupPipeline): (WebCore::GStreamerCapturer::interrupted const): (WebCore::GStreamerCapturer::setInterrupted): * platform/mediastream/gstreamer/GStreamerCapturer.h: * platform/mediastream/gstreamer/MockRealtimeAudioSourceGStreamer.cpp: (WebCore::MockRealtimeAudioSourceGStreamer::setInterruptedForTesting): * platform/mediastream/gstreamer/MockRealtimeAudioSourceGStreamer.h: LayoutTests: * platform/glib/TestExpectations: fast/mediastream/media-stream-track-interrupted.html is now passing. Canonical link: https://commits.webkit.org/236397@main git-svn-id: https://svn.webkit.org/repository/webkit/trunk@275827 268f45cc-cd09-0410-ab3c-d52691b4dbfc
diff --git a/LayoutTests/ChangeLog b/LayoutTests/ChangeLog
@@ -1,3 +1,12 @@
+2021-04-12  Philippe Normand  <pnormand@igalia.com>
+
+        [GStreamer][WebRTC] An audio track should be muted when capture is interrupted by the OS.
+        https://bugs.webkit.org/show_bug.cgi?id=196606
+
+        Reviewed by Xabier Rodriguez-Calvar.
+
+        * platform/glib/TestExpectations: fast/mediastream/media-stream-track-interrupted.html is now passing.
+
 2021-04-12  Sam Weinig  <weinig@apple.com>
 
         Update color-contrast() to support a target contrast ratio
diff --git a/LayoutTests/platform/glib/TestExpectations b/LayoutTests/platform/glib/TestExpectations
@@ -532,7 +532,6 @@ webkit.org/b/210337 fast/mediastream/mediastreamtrack-audio-clone.html [ Failure
 webkit.org/b/210385 fast/mediastream/stream-switch.html [ Crash Timeout ]
 webkit.org/b/210528 fast/mediastream/MediaStream-MediaElement-setObject-null.html [ Crash Pass ]
 webkit.org/b/210528 fast/mediastream/mediastreamtrack-video-frameRate-clone-decreasing.html [ Crash Pass ]
-webkit.org/b/213011 fast/mediastream/media-stream-track-interrupted.html [ Failure ]
 
 webkit.org/b/203078 media/media-source/media-source-remove-unload-crash.html [ Crash Timeout Pass ]
 webkit.org/b/210528 media/media-source/media-source-seek-back.html [ Crash Pass ]
diff --git a/Source/WebCore/ChangeLog b/Source/WebCore/ChangeLog
@@ -1,3 +1,25 @@
+2021-04-12  Philippe Normand  <pnormand@igalia.com>
+
+        [GStreamer][WebRTC] An audio track should be muted when capture is interrupted by the OS.
+        https://bugs.webkit.org/show_bug.cgi?id=196606
+
+        Reviewed by Xabier Rodriguez-Calvar.
+
+        Add basic interruption support in the audio capture source and mock audio capture source.
+
+        * platform/mediastream/gstreamer/GStreamerAudioCaptureSource.cpp:
+        (WebCore::GStreamerAudioCaptureSource::interrupted const):
+        (WebCore::GStreamerAudioCaptureSource::setInterruptedForTesting):
+        * platform/mediastream/gstreamer/GStreamerAudioCaptureSource.h:
+        * platform/mediastream/gstreamer/GStreamerCapturer.cpp:
+        (WebCore::GStreamerCapturer::setupPipeline):
+        (WebCore::GStreamerCapturer::interrupted const):
+        (WebCore::GStreamerCapturer::setInterrupted):
+        * platform/mediastream/gstreamer/GStreamerCapturer.h:
+        * platform/mediastream/gstreamer/MockRealtimeAudioSourceGStreamer.cpp:
+        (WebCore::MockRealtimeAudioSourceGStreamer::setInterruptedForTesting):
+        * platform/mediastream/gstreamer/MockRealtimeAudioSourceGStreamer.h:
+
 2021-04-12  Sam Weinig  <weinig@apple.com>
 
         Update color-contrast() to support a target contrast ratio
diff --git a/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCaptureSource.cpp b/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCaptureSource.cpp
@@ -208,6 +208,17 @@ const RealtimeMediaSourceSettings& GStreamerAudioCaptureSource::settings()
     return m_currentSettings.value();
 }
 
+bool GStreamerAudioCaptureSource::interrupted() const
+{
+    return m_capturer->isInterrupted() || RealtimeMediaSource::interrupted();
+}
+
+void GStreamerAudioCaptureSource::setInterruptedForTesting(bool isInterrupted)
+{
+    m_capturer->setInterrupted(isInterrupted);
+    RealtimeMediaSource::setInterruptedForTesting(isInterrupted);
+}
+
 } // namespace WebCore
 
 #endif // ENABLE(MEDIA_STREAM) && USE(GSTREAMER)
diff --git a/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCaptureSource.h b/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCaptureSource.h
@@ -53,6 +53,9 @@ class GStreamerAudioCaptureSource : public RealtimeMediaSource {
     mutable Optional<RealtimeMediaSourceSettings> m_currentSettings;
 
 private:
+    bool interrupted() const final;
+    void setInterruptedForTesting(bool) final;
+
     bool isCaptureSource() const final { return true; }
     void settingsDidChange(OptionSet<RealtimeMediaSourceSettings::Flag>) final;
 
diff --git a/Source/WebCore/platform/mediastream/gstreamer/GStreamerCapturer.cpp b/Source/WebCore/platform/mediastream/gstreamer/GStreamerCapturer.cpp
@@ -108,15 +108,16 @@ void GStreamerCapturer::setupPipeline()
     GRefPtr<GstElement> source = createSource();
     GRefPtr<GstElement> converter = createConverter();
 
+    m_valve = makeElement("valve");
     m_capsfilter = makeElement("capsfilter");
     m_tee = makeElement("tee");
     m_sink = makeElement("appsink");
 
     gst_app_sink_set_emit_signals(GST_APP_SINK(m_sink.get()), TRUE);
     g_object_set(m_capsfilter.get(), "caps", m_caps.get(), nullptr);
 
-    gst_bin_add_many(GST_BIN(m_pipeline.get()), source.get(), converter.get(), m_capsfilter.get(), m_tee.get(), nullptr);
-    gst_element_link_many(source.get(), converter.get(), m_capsfilter.get(), m_tee.get(), nullptr);
+    gst_bin_add_many(GST_BIN(m_pipeline.get()), source.get(), converter.get(), m_capsfilter.get(), m_valve.get(), m_tee.get(), nullptr);
+    gst_element_link_many(source.get(), converter.get(), m_capsfilter.get(), m_valve.get(), m_tee.get(), nullptr);
 
     addSink(m_sink.get());
 
@@ -182,6 +183,18 @@ void GStreamerCapturer::stop()
     gst_element_set_state(pipeline(), GST_STATE_NULL);
 }
 
+bool GStreamerCapturer::isInterrupted() const
+{
+    gboolean isInterrupted;
+    g_object_get(m_valve.get(), "drop", &isInterrupted, nullptr);
+    return isInterrupted;
+}
+
+void GStreamerCapturer::setInterrupted(bool isInterrupted)
+{
+    g_object_set(m_valve.get(), "drop", isInterrupted, nullptr);
+}
+
 } // namespace WebCore
 
 #endif // ENABLE(VIDEO) && ENABLE(MEDIA_STREAM) && USE(GSTREAMER)
diff --git a/Source/WebCore/platform/mediastream/gstreamer/GStreamerCapturer.h b/Source/WebCore/platform/mediastream/gstreamer/GStreamerCapturer.h
@@ -39,8 +39,8 @@ class GStreamerCapturer {
     ~GStreamerCapturer();
 
     void setupPipeline();
-    virtual void play();
-    virtual void stop();
+    void play();
+    void stop();
     GstCaps* caps();
     void addSink(GstElement *newSink);
     GstElement* makeElement(const char* factoryName);
@@ -54,9 +54,13 @@ class GStreamerCapturer {
     GstElement* pipeline() const { return m_pipeline.get(); }
     virtual GstElement* createConverter() = 0;
 
+    bool isInterrupted() const;
+    void setInterrupted(bool);
+
 protected:
     GRefPtr<GstElement> m_sink;
     GRefPtr<GstElement> m_src;
+    GRefPtr<GstElement> m_valve;
     GRefPtr<GstElement> m_tee;
     GRefPtr<GstElement> m_capsfilter;
     GRefPtr<GstDevice> m_device;
diff --git a/Source/WebCore/platform/mediastream/gstreamer/MockRealtimeAudioSourceGStreamer.cpp b/Source/WebCore/platform/mediastream/gstreamer/MockRealtimeAudioSourceGStreamer.cpp
@@ -140,6 +140,12 @@ void MockRealtimeAudioSourceGStreamer::reconfigure()
         addHum(s_NoiseVolume, s_NoiseFrequency, rate, 0, m_bipBopBuffer.data(), sampleCount);
 }
 
+void MockRealtimeAudioSourceGStreamer::setInterruptedForTesting(bool isInterrupted)
+{
+    m_isInterrupted = isInterrupted;
+    MockRealtimeAudioSource::setInterruptedForTesting(isInterrupted);
+}
+
 } // namespace WebCore
 
 #endif // ENABLE(MEDIA_STREAM) && USE(GSTREAMER)
diff --git a/Source/WebCore/platform/mediastream/gstreamer/MockRealtimeAudioSourceGStreamer.h b/Source/WebCore/platform/mediastream/gstreamer/MockRealtimeAudioSourceGStreamer.h
@@ -45,11 +45,15 @@ class MockRealtimeAudioSourceGStreamer final : public MockRealtimeAudioSource {
     void reconfigure();
     void addHum(float amplitude, float frequency, float sampleRate, uint64_t start, float *p, uint64_t count);
 
+    bool interrupted() const final { return m_isInterrupted; };
+    void setInterruptedForTesting(bool) final;
+
     Optional<GStreamerAudioStreamDescription> m_streamFormat;
     Vector<float> m_bipBopBuffer;
     uint32_t m_maximiumFrameCount;
     uint64_t m_samplesEmitted { 0 };
     uint64_t m_samplesRendered { 0 };
+    bool m_isInterrupted { false };
 };
 
 } // namespace WebCore
