Title: [277016] trunk/Source/WebCore
- Revision
- 277016
- Author
- commit-qu...@webkit.org
- Date
- 2021-05-05 06:55:12 -0700 (Wed, 05 May 2021)
Log Message
[GStreamer][WebRTC] Wrap incoming audio buffers, complying with fastMalloc restrictions
https://bugs.webkit.org/show_bug.cgi?id=225351
Patch by Philippe Normand <pnorm...@igalia.com> on 2021-05-05
Reviewed by Xabier Rodriguez-Calvar.
The audio thread should avoid calling fastMalloc for performance reasons, so audio buffers
are now wrapped in GStreamer buffers and passed as read-only memories to the observers.
* platform/mediastream/gstreamer/RealtimeIncomingAudioSourceLibWebRTC.cpp:
(WebCore::RealtimeIncomingAudioSourceLibWebRTC::OnData):
Modified Paths
Diff
Modified: trunk/Source/WebCore/ChangeLog (277015 => 277016)
--- trunk/Source/WebCore/ChangeLog 2021-05-05 13:39:11 UTC (rev 277015)
+++ trunk/Source/WebCore/ChangeLog 2021-05-05 13:55:12 UTC (rev 277016)
@@ -1,3 +1,16 @@
+2021-05-05 Philippe Normand <pnorm...@igalia.com>
+
+ [GStreamer][WebRTC] Wrap incoming audio buffers, complying with fastMalloc restrictions
+ https://bugs.webkit.org/show_bug.cgi?id=225351
+
+ Reviewed by Xabier Rodriguez-Calvar.
+
+ The audio thread should avoid calling fastMalloc for performance reasons, so audio buffers
+ are now wrapped in GStreamer buffers and passed as read-only memories to the observers.
+
+ * platform/mediastream/gstreamer/RealtimeIncomingAudioSourceLibWebRTC.cpp:
+ (WebCore::RealtimeIncomingAudioSourceLibWebRTC::OnData):
+
2021-05-05 Carlos Garcia Campos <cgar...@igalia.com>
[SOUP] Wrong cookie timestamp in case of long expire time
Modified: trunk/Source/WebCore/platform/mediastream/gstreamer/RealtimeIncomingAudioSourceLibWebRTC.cpp (277015 => 277016)
--- trunk/Source/WebCore/platform/mediastream/gstreamer/RealtimeIncomingAudioSourceLibWebRTC.cpp 2021-05-05 13:39:11 UTC (rev 277015)
+++ trunk/Source/WebCore/platform/mediastream/gstreamer/RealtimeIncomingAudioSourceLibWebRTC.cpp 2021-05-05 13:55:12 UTC (rev 277016)
@@ -64,17 +64,8 @@
gst_audio_info_set_format(&info, format, sampleRate, numberOfChannels, NULL);
- // FIXME: We could likely avoid allocation here, when muted -> memset(0...) and maybe wrap the
- // audioData in a GstBuffer?
- DisableMallocRestrictionsForCurrentThreadScope disableMallocRestrictions;
auto bufferSize = GST_AUDIO_INFO_BPF(&info) * numberOfFrames;
- gpointer bufferData = fastMalloc(bufferSize);
- if (muted())
- webkitGstAudioFormatFillSilence(info.finfo, bufferData, bufferSize);
- else
- memcpy(bufferData, audioData, bufferSize);
-
- auto buffer = adoptGRef(gstBufferNewWrappedFast(bufferData, bufferSize));
+ auto buffer = adoptGRef(gst_buffer_new_wrapped_full(GST_MEMORY_FLAG_READONLY, const_cast<gpointer>(audioData), bufferSize, 0, bufferSize, nullptr, nullptr));
auto caps = adoptGRef(gst_audio_info_to_caps(&info));
auto sample = adoptGRef(gst_sample_new(buffer.get(), caps.get(), nullptr, nullptr));
GStreamerAudioData data(WTFMove(sample), info);
_______________________________________________
webkit-changes mailing list
webkit-changes@lists.webkit.org
https://lists.webkit.org/mailman/listinfo/webkit-changes
