Diff
Modified: trunk/Source/WebCore/ChangeLog (274786 => 274787)
--- trunk/Source/WebCore/ChangeLog 2021-03-22 20:15:04 UTC (rev 274786)
+++ trunk/Source/WebCore/ChangeLog 2021-03-22 20:22:14 UTC (rev 274787)
@@ -1,3 +1,20 @@
+2021-03-22 Philippe Normand <[email protected]>
+
+ [GStreamer] gst_audio_format_fill_silence() is deprecated in GStreamer 1.20
+ https://bugs.webkit.org/show_bug.cgi?id=223562
+
+ Reviewed by Xabier Rodriguez-Calvar.
+
+ Fix GStreamer deprecation warnings.
+
+ * platform/graphics/gstreamer/GStreamerCommon.h:
+ * platform/mediastream/gstreamer/MockRealtimeAudioSourceGStreamer.cpp:
+ (WebCore::MockRealtimeAudioSourceGStreamer::render):
+ * platform/mediastream/gstreamer/RealtimeIncomingAudioSourceLibWebRTC.cpp:
+ (WebCore::RealtimeIncomingAudioSourceLibWebRTC::OnData):
+ * platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp:
+ (WebCore::RealtimeOutgoingAudioSourceLibWebRTC::pullAudioData):
+
2021-03-22 Youenn Fablet <[email protected]>
Async Clipboard read prevents WebRTC IOSurfaces from being released
Modified: trunk/Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.h (274786 => 274787)
--- trunk/Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.h 2021-03-22 20:15:04 UTC (rev 274786)
+++ trunk/Source/WebCore/platform/graphics/gstreamer/GStreamerCommon.h 2021-03-22 20:22:14 UTC (rev 274787)
@@ -307,4 +307,13 @@
#ifndef GST_BUFFER_DTS_OR_PTS
#define GST_BUFFER_DTS_OR_PTS(buffer) (GST_BUFFER_DTS_IS_VALID(buffer) ? GST_BUFFER_DTS(buffer) : GST_BUFFER_PTS(buffer))
#endif
+
+// In GStreamer 1.20 gst_audio_format_fill_silence() will be deprecated in favor of
+// gst_audio_format_info_fill_silence().
+#if GST_CHECK_VERSION(1, 19, 0)
+#define webkitGstAudioFormatFillSilence gst_audio_format_info_fill_silence
+#else
+#define webkitGstAudioFormatFillSilence gst_audio_format_fill_silence
+#endif
+
#endif // USE(GSTREAMER)
Modified: trunk/Source/WebCore/platform/mediastream/gstreamer/MockRealtimeAudioSourceGStreamer.cpp (274786 => 274787)
--- trunk/Source/WebCore/platform/mediastream/gstreamer/MockRealtimeAudioSourceGStreamer.cpp 2021-03-22 20:15:04 UTC (rev 274786)
+++ trunk/Source/WebCore/platform/mediastream/gstreamer/MockRealtimeAudioSourceGStreamer.cpp 2021-03-22 20:22:14 UTC (rev 274787)
@@ -88,7 +88,7 @@
GstMappedBuffer map(buffer.get(), GST_MAP_WRITE);
if (muted())
- gst_audio_format_fill_silence(info.finfo, map.data(), map.size());
+ webkitGstAudioFormatFillSilence(info.finfo, map.data(), map.size());
else {
memcpy(map.data(), &m_bipBopBuffer[bipBopStart], sizeof(float) * bipBopCount);
addHum(s_HumVolume, s_HumFrequency, sampleRate(), m_samplesRendered, reinterpret_cast<float*>(map.data()), bipBopCount);
Modified: trunk/Source/WebCore/platform/mediastream/gstreamer/RealtimeIncomingAudioSourceLibWebRTC.cpp (274786 => 274787)
--- trunk/Source/WebCore/platform/mediastream/gstreamer/RealtimeIncomingAudioSourceLibWebRTC.cpp 2021-03-22 20:15:04 UTC (rev 274786)
+++ trunk/Source/WebCore/platform/mediastream/gstreamer/RealtimeIncomingAudioSourceLibWebRTC.cpp 2021-03-22 20:22:14 UTC (rev 274787)
@@ -67,7 +67,7 @@
auto bufferSize = GST_AUDIO_INFO_BPF(&info) * numberOfFrames;
gpointer bufferData = fastMalloc(bufferSize);
if (muted())
- gst_audio_format_fill_silence(info.finfo, bufferData, bufferSize);
+ webkitGstAudioFormatFillSilence(info.finfo, bufferData, bufferSize);
else
memcpy(bufferData, audioData, bufferSize);
Modified: trunk/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp (274786 => 274787)
--- trunk/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp 2021-03-22 20:15:04 UTC (rev 274786)
+++ trunk/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp 2021-03-22 20:22:14 UTC (rev 274787)
@@ -108,7 +108,7 @@
auto inBuffer = adoptGRef(gst_adapter_take_buffer(m_adapter.get(), inBufferSize));
m_audioBuffer.grow(outBufferSize);
if (isSilenced())
- gst_audio_format_fill_silence(m_outputStreamDescription.finfo, m_audioBuffer.data(), outBufferSize);
+ webkitGstAudioFormatFillSilence(m_outputStreamDescription.finfo, m_audioBuffer.data(), outBufferSize);
else {
GstMappedBuffer inMap(inBuffer.get(), GST_MAP_READ);
