Modified: trunk/Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.cpp (268578 => 268579)
--- trunk/Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.cpp 2020-10-16 10:55:59 UTC (rev 268578)
+++ trunk/Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.cpp 2020-10-16 11:22:06 UTC (rev 268579)
@@ -87,11 +87,8 @@
gst_bus_add_signal_watch_full(bus.get(), RunLoopSourcePriority::RunLoopDispatcher);
g_signal_connect(bus.get(), "message", G_CALLBACK(messageCallback), this);
- GstElement* webkitAudioSrc = reinterpret_cast<GstElement*>(g_object_new(WEBKIT_TYPE_WEB_AUDIO_SRC,
- "rate", sampleRate,
- "bus", m_renderBus.get(),
- "provider", &m_callback,
- "frames", AudioUtilities::renderQuantumSize, nullptr));
+ m_src = reinterpret_cast<GstElement*>(g_object_new(WEBKIT_TYPE_WEB_AUDIO_SRC, "rate", sampleRate,
+ "bus", m_renderBus.get(), "provider", &m_callback, "frames", AudioUtilities::renderQuantumSize, nullptr));
GRefPtr<GstElement> audioSink = createPlatformAudioSink();
m_audioSinkAvailable = audioSink;
@@ -119,10 +116,10 @@
GstElement* audioConvert = gst_element_factory_make("audioconvert", nullptr);
GstElement* audioResample = gst_element_factory_make("audioresample", nullptr);
- gst_bin_add_many(GST_BIN_CAST(m_pipeline), webkitAudioSrc, audioConvert, audioResample, audioSink.get(), nullptr);
+ gst_bin_add_many(GST_BIN_CAST(m_pipeline), m_src.get(), audioConvert, audioResample, audioSink.get(), nullptr);
// Link src pads from webkitAudioSrc to audioConvert ! audioResample ! autoaudiosink.
- gst_element_link_pads_full(webkitAudioSrc, "src", audioConvert, "sink", GST_PAD_LINK_CHECK_NOTHING);
+ gst_element_link_pads_full(m_src.get(), "src", audioConvert, "sink", GST_PAD_LINK_CHECK_NOTHING);
gst_element_link_pads_full(audioConvert, "src", audioResample, "sink", GST_PAD_LINK_CHECK_NOTHING);
gst_element_link_pads_full(audioResample, "src", audioSink.get(), "sink", GST_PAD_LINK_CHECK_NOTHING);
}
@@ -165,12 +162,15 @@
return TRUE;
}
-void AudioDestinationGStreamer::start(Function<void(Function<void()>&&)>&&)
+void AudioDestinationGStreamer::start(Function<void(Function<void()>&&)>&& dispatchToRenderThread)
{
ASSERT(m_audioSinkAvailable);
if (!m_audioSinkAvailable)
return;
+ if (dispatchToRenderThread)
+ webkitWebAudioSourceSetDispatchToRenderThreadCallback(WEBKIT_WEB_AUDIO_SRC(m_src.get()), WTFMove(dispatchToRenderThread));
+
if (gst_element_set_state(m_pipeline, GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE) {
g_warning("Error: Failed to set pipeline to playing");
m_isPlaying = false;
Modified: trunk/Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.h (268578 => 268579)
--- trunk/Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.h 2020-10-16 10:55:59 UTC (rev 268578)
+++ trunk/Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.h 2020-10-16 11:22:06 UTC (rev 268579)
@@ -21,7 +21,8 @@
#include "AudioBus.h"
#include "AudioDestination.h"
-#include <wtf/RefPtr.h>
+#include "GRefPtrGStreamer.h"
+#include <wtf/Forward.h>
typedef struct _GstElement GstElement;
typedef struct _GstPad GstPad;
@@ -52,6 +53,7 @@
bool m_isPlaying;
bool m_audioSinkAvailable;
GstElement* m_pipeline;
+ GRefPtr<GstElement> m_src;
};
} // namespace WebCore
Modified: trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp (268578 => 268579)
--- trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp 2020-10-16 10:55:59 UTC (rev 268578)
+++ trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp 2020-10-16 11:22:06 UTC (rev 268579)
@@ -30,6 +30,8 @@
#include <gst/app/gstappsrc.h>
#include <gst/audio/audio-info.h>
#include <gst/pbutils/missing-plugins.h>
+#include <wtf/Condition.h>
+#include <wtf/Lock.h>
#include <wtf/glib/GUniquePtr.h>
#include <wtf/glib/WTFGType.h>
@@ -77,6 +79,10 @@
bool enableGapBufferSupport;
+ Optional<Function<void(Function<void()>&&)>> dispatchToRenderThreadCallback;
+ Lock dispatchMutex;
+ Condition dispatchCondition;
+
_WebKitWebAudioSrcPrivate()
{
sourcePad = webkitGstGhostPadFromStaticTemplate(&srcTemplate, "src", nullptr);
@@ -340,8 +346,18 @@
}
// FIXME: Add support for local/live audio input.
- priv->provider->render(nullptr, priv->bus, priv->framesToPull, outputTimestamp);
+ if (src->priv->dispatchToRenderThreadCallback.hasValue()) {
+ LockHolder holder(priv->dispatchMutex);
+ (*priv->dispatchToRenderThreadCallback)([src, outputTimestamp]() mutable {
+ auto* priv = src->priv;
+ priv->provider->render(nullptr, priv->bus, priv->framesToPull, outputTimestamp);
+ priv->dispatchCondition.notifyOne();
+ });
+ priv->dispatchCondition.wait(priv->dispatchMutex);
+ } else
+ priv->provider->render(nullptr, priv->bus, priv->framesToPull, outputTimestamp);
+
return makeOptional(channelBufferList);
}
@@ -434,4 +450,10 @@
return returnValue;
}
+void webkitWebAudioSourceSetDispatchToRenderThreadCallback(WebKitWebAudioSrc* src, Function<void(Function<void()>&&)>&& function)
+{
+ ASSERT(function);
+ src->priv->dispatchToRenderThreadCallback = WTFMove(function);
+}
+
#endif // ENABLE(WEB_AUDIO) && USE(GSTREAMER)
Modified: trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.h (268578 => 268579)
--- trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.h 2020-10-16 10:55:59 UTC (rev 268578)
+++ trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.h 2020-10-16 11:22:06 UTC (rev 268579)
@@ -16,11 +16,12 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-#ifndef WebKitWebAudioSourceGStreamer_h
-#define WebKitWebAudioSourceGStreamer_h
+#pragma once
+
#if USE(GSTREAMER)
#include <gst/gst.h>
+#include <wtf/Forward.h>
#define WEBKIT_TYPE_WEB_AUDIO_SRC (webkit_web_audio_src_get_type())
#define WEBKIT_WEB_AUDIO_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj), WEBKIT_TYPE_WEB_AUDIO_SRC, WebKitWebAudioSrc))
@@ -29,5 +30,7 @@
GType webkit_web_audio_src_get_type();
+void webkitWebAudioSourceSetDispatchToRenderThreadCallback(WebKitWebAudioSrc*, Function<void(Function<void()>&&)>&&);
+
#endif // USE(GSTREAMER)
-#endif
+
