[CherryPick] g_slist_reverse() may not be required in webKitWebAudioSrcLoop

[Title] [CherryPick] g_slist_reverse() may not be required in webKitWebAudioSrcLoop
[Issues] N/A
[Problem] WebAudio performance optimization patch
[Solution] Cherry picked.
[Cherry-Picker] Praveen R Jadhav <praveen.j@samsung.com>

g_slist_reverse() may not be required in webKitWebAudioSrcLoop
https://bugs.webkit.org/show_bug.cgi?id=113568

Patch by Praveen R Jadhav <praveen.j@samsung.com> on 2013-03-30
Reviewed by Philippe Normand.

Decremental 'for' loop logic implemented to avoid using g_slist_reverse().

Original code            - 2.025230 micro seconds per loop
Original code + patch    - 1.964759 micro seconds per loop

This patch is covered by existing webaudio tests.

* platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp:
(webKitWebAudioSrcLoop):

git-svn-id: http://svn.webkit.org/repository/webkit/trunk@147278 268f45cc-cd09-0410-ab3c-d52691b4dbfc

Conflicts:

	Source/WebCore/ChangeLog

Change-Id: Iaba2cef05405b383adba5dc9f9e11f8eea72f704

diff --git a/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp b/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp
index 33f359c..72911b7 100755 (executable)
--- a/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp
+++ b/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp
@@ -355,7 +355,7 @@ static void webKitWebAudioSrcLoop(WebKitWebAudioSrc* src)
 
     GSList* channelBufferList = 0;
     unsigned bufferSize = priv->framesToPull * sizeof(float);
-    for (unsigned i = 0; i < g_slist_length(priv->pads); i++) {
+    for (int 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);
@@ -368,12 +368,11 @@ static void webKitWebAudioSrcLoop(WebKitWebAudioSrc* src)
         priv->bus->setChannelMemory(i, reinterpret_cast<float*>(GST_BUFFER_DATA(channelBuffer)), priv->framesToPull);
 #endif
     }
-    channelBufferList = g_slist_reverse(channelBufferList);
 
     // FIXME: Add support for local/live audio input.
     priv->provider->render(0, priv->bus, priv->framesToPull);
 
-    for (unsigned i = 0; i < g_slist_length(priv->pads); i++) {
+    for (int i = g_slist_length(priv->pads) - 1; i >= 0; i--) {
         GstPad* pad = static_cast<GstPad*>(g_slist_nth_data(priv->pads, i));
         GstBuffer* channelBuffer = static_cast<GstBuffer*>(g_slist_nth_data(channelBufferList, i));