From 9044d10efcccc74c8645e66be3b54142a8ae0b1b Mon Sep 17 00:00:00 2001 From: "philn@webkit.org" Date: Mon, 23 Jan 2012 16:24:21 +0000 Subject: [PATCH] [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 --- Source/WebCore/ChangeLog | 14 ++++++++++++++ .../platform/audio/gstreamer/AudioFileReaderGStreamer.cpp | 2 +- .../audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp | 2 +- 3 files changed, 16 insertions(+), 2 deletions(-) diff --git a/Source/WebCore/ChangeLog b/Source/WebCore/ChangeLog index a87f23a..95fc716 100644 --- a/Source/WebCore/ChangeLog +++ b/Source/WebCore/ChangeLog @@ -1,3 +1,17 @@ +2012-01-23 Philippe Normand + + [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 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 --- 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(GST_BUFFER_DATA(buffer)), GST_BUFFER_SIZE(buffer)); + memcpy(audioChannel->mutableData(), reinterpret_cast(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 --- 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(priv->bus->channel(index)->data()); + GST_BUFFER_DATA(buffer) = reinterpret_cast(const_cast(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; -- 2.7.4