Title: [152832] trunk/Source/WebCore
- Revision
- 152832
- Author
- [email protected]
- Date
- 2013-07-18 00:27:16 -0700 (Thu, 18 Jul 2013)
Log Message
[gstreamer] Avoid calls to g_slist_index in webKitWebAudioSrcLoop()
https://bugs.webkit.org/show_bug.cgi?id=118827
Patch by Nick Diego Yamane <[email protected]> on 2013-07-18
Reviewed by Philippe Normand.
webKitWebAudioSrcLoop() currently calls g_slist_index for each element
to get its index in a list it's iterating over. g_list_index function uses
a sequential search to find that element, which is clearly unecessary.
This patch adds a local variable to store the current index and use it
instead of calling g_slist_index function.
No new tests, no behavior changes.
* platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp:
(webKitWebAudioSrcLoop):
Modified Paths
Diff
Modified: trunk/Source/WebCore/ChangeLog (152831 => 152832)
--- trunk/Source/WebCore/ChangeLog 2013-07-18 07:27:08 UTC (rev 152831)
+++ trunk/Source/WebCore/ChangeLog 2013-07-18 07:27:16 UTC (rev 152832)
@@ -1,3 +1,21 @@
+2013-07-18 Nick Diego Yamane <[email protected]>
+
+ [gstreamer] Avoid calls to g_slist_index in webKitWebAudioSrcLoop()
+ https://bugs.webkit.org/show_bug.cgi?id=118827
+
+ Reviewed by Philippe Normand.
+
+ webKitWebAudioSrcLoop() currently calls g_slist_index for each element
+ to get its index in a list it's iterating over. g_list_index function uses
+ a sequential search to find that element, which is clearly unecessary.
+ This patch adds a local variable to store the current index and use it
+ instead of calling g_slist_index function.
+
+ No new tests, no behavior changes.
+
+ * platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp:
+ (webKitWebAudioSrcLoop):
+
2013-07-17 Gyuyoung Kim <[email protected]>
Introduce toSVGInlineFlowBox() and use it
Modified: trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp (152831 => 152832)
--- trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp 2013-07-18 07:27:08 UTC (rev 152831)
+++ trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp 2013-07-18 07:27:16 UTC (rev 152832)
@@ -354,8 +354,9 @@
return;
GSList* channelBufferList = 0;
+ register int i;
unsigned bufferSize = priv->framesToPull * sizeof(float);
- for (int i = g_slist_length(priv->pads) - 1; i >= 0; i--) {
+ for (i = g_slist_length(priv->pads) - 1; i >= 0; i--) {
GstBuffer* channelBuffer = gst_buffer_new_and_alloc(bufferSize);
ASSERT(channelBuffer);
channelBufferList = g_slist_prepend(channelBufferList, channelBuffer);
@@ -372,16 +373,16 @@
// FIXME: Add support for local/live audio input.
priv->provider->render(0, priv->bus, priv->framesToPull);
- GSList* padsIt;
- GSList* buffersIt;
- for (padsIt = priv->pads, buffersIt = channelBufferList; padsIt && buffersIt; padsIt = g_slist_next(padsIt), buffersIt = g_slist_next(buffersIt)) {
+ GSList* padsIt = priv->pads;
+ GSList* buffersIt = channelBufferList;
+ for (i = 0; padsIt && buffersIt; padsIt = g_slist_next(padsIt), buffersIt = g_slist_next(buffersIt), ++i) {
GstPad* pad = static_cast<GstPad*>(padsIt->data);
GstBuffer* channelBuffer = static_cast<GstBuffer*>(buffersIt->data);
#ifndef GST_API_VERSION_1
GRefPtr<GstCaps> monoCaps = adoptGRef(getGStreamerMonoAudioCaps(priv->sampleRate));
GstStructure* structure = gst_caps_get_structure(monoCaps.get(), 0);
- GstAudioChannelPosition channelPosition = webKitWebAudioGStreamerChannelPosition(g_slist_index(channelBufferList, channelBuffer));
+ GstAudioChannelPosition channelPosition = webKitWebAudioGStreamerChannelPosition(i);
gst_audio_set_channel_positions(structure, &channelPosition);
gst_buffer_set_caps(channelBuffer, monoCaps.get());
#endif
_______________________________________________
webkit-changes mailing list
[email protected]
https://lists.webkit.org/mailman/listinfo/webkit-changes
