[GStreamer] fix WebAudio build after r105431

https://bugs.webkit.org/show_bug.cgi?id=76819

Reviewed by Martin Robinson.

* platform/audio/gstreamer/AudioFileReaderGStreamer.cpp:
(WebCore::copyGstreamerBuffersToAudioChannel): Use mutableData()
when copying.
* platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp:
(webKitWebAudioSrcLoop): Drop constness when setting the buffer
data pointer.

git-svn-id: http://svn.webkit.org/repository/webkit/trunk@105626 268f45cc-cd09-0410-ab3c-d52691b4dbfc

diff --git a/Source/WebCore/ChangeLog b/Source/WebCore/ChangeLog
index a87f23a..95fc716 100644 (file)
--- a/Source/WebCore/ChangeLog
+++ b/Source/WebCore/ChangeLog
@@ -1,3 +1,17 @@
+2012-01-23  Philippe Normand  <pnormand@igalia.com>
+
+        [GStreamer] fix WebAudio build after r105431
+        https://bugs.webkit.org/show_bug.cgi?id=76819
+
+        Reviewed by Martin Robinson.
+
+        * platform/audio/gstreamer/AudioFileReaderGStreamer.cpp:
+        (WebCore::copyGstreamerBuffersToAudioChannel): Use mutableData()
+        when copying.
+        * platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp:
+        (webKitWebAudioSrcLoop): Drop constness when setting the buffer
+        data pointer.
+
 2012-01-23  Pavel Feldman  <pfeldman@google.com>
 
         Web Inspector: add touch events to the event listeners list.

diff --git a/Source/WebCore/platform/audio/gstreamer/AudioFileReaderGStreamer.cpp b/Source/WebCore/platform/audio/gstreamer/AudioFileReaderGStreamer.cpp
index 20e81b9..093f81f 100644 (file)
--- a/Source/WebCore/platform/audio/gstreamer/AudioFileReaderGStreamer.cpp
+++ b/Source/WebCore/platform/audio/gstreamer/AudioFileReaderGStreamer.cpp
@@ -80,7 +80,7 @@ static void copyGstreamerBuffersToAudioChannel(GstBufferList* buffers, AudioChan
     gst_buffer_list_iterator_next_group(iter);
     GstBuffer* buffer = gst_buffer_list_iterator_merge_group(iter);
     if (buffer) {
-        memcpy(audioChannel->data(), reinterpret_cast<float*>(GST_BUFFER_DATA(buffer)), GST_BUFFER_SIZE(buffer));
+        memcpy(audioChannel->mutableData(), reinterpret_cast<float*>(GST_BUFFER_DATA(buffer)), GST_BUFFER_SIZE(buffer));
         gst_buffer_unref(buffer);
     }
 

diff --git a/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp b/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp
index 5183e34..2cae4ac 100644 (file)
--- a/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp
+++ b/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp
@@ -333,7 +333,7 @@ static void webKitWebAudioSrcLoop(WebKitWebAudioSrc* src)
         ASSERT(buffer);
         ASSERT(!GST_BUFFER_MALLOCDATA(buffer));
 
-        GST_BUFFER_DATA(buffer) = reinterpret_cast<guint8*>(priv->bus->channel(index)->data());
+        GST_BUFFER_DATA(buffer) = reinterpret_cast<guint8*>(const_cast<float*>(priv->bus->channel(index)->data()));
         GST_BUFFER_SIZE(buffer) = bufferSize;
         GST_BUFFER_OFFSET(buffer) = priv->currentBufferOffset;
         GST_BUFFER_OFFSET_END(buffer) = priv->currentBufferOffset + priv->framesToPull;