Diff
Modified: releases/WebKitGTK/webkit-2.16/Source/WebCore/ChangeLog (213812 => 213813)
--- releases/WebKitGTK/webkit-2.16/Source/WebCore/ChangeLog 2017-03-13 10:40:05 UTC (rev 213812)
+++ releases/WebKitGTK/webkit-2.16/Source/WebCore/ChangeLog 2017-03-13 10:41:21 UTC (rev 213813)
@@ -1,3 +1,91 @@
+2017-03-06 Vanessa Chipirrás Navalón <[email protected]>
+
+ [GStreamer] Adopt nullptr
+ https://bugs.webkit.org/show_bug.cgi?id=123438
+
+ Reviewed by Xabier Rodriguez-Calvar.
+
+ To adapt the code to the C++11 standard, all NULL or 0 pointers have been changed to nullptr.
+
+ * platform/audio/gstreamer/AudioDestinationGStreamer.cpp:
+ (WebCore::AudioDestinationGStreamer::AudioDestinationGStreamer):
+ * platform/audio/gstreamer/AudioFileReaderGStreamer.cpp:
+ (WebCore::AudioFileReader::handleNewDeinterleavePad):
+ (WebCore::AudioFileReader::plugDeinterleave):
+ (WebCore::AudioFileReader::decodeAudioForBusCreation):
+ * platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp:
+ (WebCore::AudioSourceProviderGStreamer::AudioSourceProviderGStreamer):
+ (WebCore::AudioSourceProviderGStreamer::configureAudioBin):
+ (WebCore::AudioSourceProviderGStreamer::setClient):
+ (WebCore::AudioSourceProviderGStreamer::handleNewDeinterleavePad):
+ * platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp:
+ (webkit_web_audio_src_init):
+ (webKitWebAudioSrcLoop):
+ (webKitWebAudioSrcChangeState):
+ * platform/graphics/gstreamer/AudioTrackPrivateGStreamer.cpp:
+ (WebCore::AudioTrackPrivateGStreamer::setEnabled):
+ * platform/graphics/gstreamer/GStreamerUtilities.cpp:
+ (WebCore::initializeGStreamer):
+ * platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp:
+ (WebCore::MediaPlayerPrivateGStreamer::setAudioStreamProperties):
+ (WebCore::MediaPlayerPrivateGStreamer::registerMediaEngine):
+ (WebCore::initializeGStreamerAndRegisterWebKitElements):
+ (WebCore::MediaPlayerPrivateGStreamer::MediaPlayerPrivateGStreamer):
+ (WebCore::MediaPlayerPrivateGStreamer::~MediaPlayerPrivateGStreamer):
+ (WebCore::MediaPlayerPrivateGStreamer::newTextSample):
+ (WebCore::MediaPlayerPrivateGStreamer::handleMessage):
+ (WebCore::MediaPlayerPrivateGStreamer::processTableOfContents):
+ Removed the unused second argument on processTableOfContentsEntry function.
+ (WebCore::MediaPlayerPrivateGStreamer::processTableOfContentsEntry):
+ Removed the unused second argument on this function.
+ (WebCore::MediaPlayerPrivateGStreamer::fillTimerFired):
+ (WebCore::MediaPlayerPrivateGStreamer::loadNextLocation):
+ (WebCore::MediaPlayerPrivateGStreamer::createAudioSink):
+ (WebCore::MediaPlayerPrivateGStreamer::createGSTPlayBin):
+ * platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.h:
+ Removed the unused second argument on processTableOfContentsEntry function.
+ * platform/graphics/gstreamer/MediaPlayerPrivateGStreamerBase.cpp:
+ (WebCore::MediaPlayerPrivateGStreamerBase::MediaPlayerPrivateGStreamerBase):
+ (WebCore::MediaPlayerPrivateGStreamerBase::setMuted):
+ (WebCore::MediaPlayerPrivateGStreamerBase::muted):
+ (WebCore::MediaPlayerPrivateGStreamerBase::notifyPlayerOfMute):
+ (WebCore::MediaPlayerPrivateGStreamerBase::setStreamVolumeElement):
+ (WebCore::MediaPlayerPrivateGStreamerBase::decodedFrameCount):
+ (WebCore::MediaPlayerPrivateGStreamerBase::droppedFrameCount):
+ * platform/graphics/gstreamer/MediaPlayerPrivateGStreamerOwr.cpp:
+ (WebCore::MediaPlayerPrivateGStreamerOwr::registerMediaEngine):
+ * platform/graphics/gstreamer/TextCombinerGStreamer.cpp:
+ (webkit_text_combiner_init):
+ (webkitTextCombinerPadEvent):
+ (webkitTextCombinerRequestNewPad):
+ (webkitTextCombinerNew):
+ * platform/graphics/gstreamer/TextSinkGStreamer.cpp:
+ (webkitTextSinkNew):
+ * platform/graphics/gstreamer/TrackPrivateBaseGStreamer.cpp:
+ (WebCore::TrackPrivateBaseGStreamer::tagsChanged):
+ (WebCore::TrackPrivateBaseGStreamer::notifyTrackOfActiveChanged):
+ * platform/graphics/gstreamer/VideoSinkGStreamer.cpp:
+ (webkit_video_sink_init):
+ (webkitVideoSinkProposeAllocation):
+ (webkitVideoSinkNew):
+ * platform/graphics/gstreamer/VideoTrackPrivateGStreamer.cpp:
+ (WebCore::VideoTrackPrivateGStreamer::setSelected):
+ * platform/graphics/gstreamer/WebKitWebSourceGStreamer.cpp:
+ (webkit_web_src_init):
+ (webKitWebSrcDispose):
+ (webKitWebSrcSetProperty):
+ (webKitWebSrcStop):
+ (webKitWebSrcChangeState):
+ (webKitWebSrcQueryWithParent):
+ (webKitWebSrcGetProtocols):
+ (StreamingClient::handleResponseReceived):
+ (StreamingClient::handleDataReceived):
+ (ResourceHandleStreamingClient::didFail):
+ (ResourceHandleStreamingClient::wasBlocked):
+ (ResourceHandleStreamingClient::cannotShowURL):
+ * platform/graphics/gstreamer/mse/WebKitMediaSourceGStreamer.cpp:
+ (webKitMediaSrcGetProtocols):
+
2017-03-05 Simon Fraser <[email protected]>
Avoid backing store for layers with empty text nodes in a few more cases
Modified: releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.cpp (213812 => 213813)
--- releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.cpp 2017-03-13 10:40:05 UTC (rev 213812)
+++ releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.cpp 2017-03-13 10:41:21 UTC (rev 213813)
@@ -92,9 +92,9 @@
"rate", sampleRate,
"bus", m_renderBus.get(),
"provider", &m_callback,
- "frames", framesToPull, NULL));
+ "frames", framesToPull, nullptr));
- GRefPtr<GstElement> audioSink = gst_element_factory_make("autoaudiosink", 0);
+ GRefPtr<GstElement> audioSink = gst_element_factory_make("autoaudiosink", nullptr);
m_audioSinkAvailable = audioSink;
if (!audioSink) {
LOG_ERROR("Failed to create GStreamer autoaudiosink element");
@@ -114,9 +114,9 @@
return;
}
- GstElement* audioConvert = gst_element_factory_make("audioconvert", 0);
- GstElement* audioResample = gst_element_factory_make("audioresample", 0);
- gst_bin_add_many(GST_BIN(m_pipeline), webkitAudioSrc, audioConvert, audioResample, audioSink.get(), NULL);
+ GstElement* audioConvert = gst_element_factory_make("audioconvert", nullptr);
+ GstElement* audioResample = gst_element_factory_make("audioresample", nullptr);
+ gst_bin_add_many(GST_BIN(m_pipeline), webkitAudioSrc, audioConvert, audioResample, audioSink.get(), nullptr);
// Link src pads from webkitAudioSrc to audioConvert ! audioResample ! autoaudiosink.
gst_element_link_pads_full(webkitAudioSrc, "src", audioConvert, "sink", GST_PAD_LINK_CHECK_NOTHING);
Modified: releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/audio/gstreamer/AudioFileReaderGStreamer.cpp (213812 => 213813)
--- releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/audio/gstreamer/AudioFileReaderGStreamer.cpp 2017-03-13 10:40:05 UTC (rev 213812)
+++ releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/audio/gstreamer/AudioFileReaderGStreamer.cpp 2017-03-13 10:41:21 UTC (rev 213813)
@@ -213,8 +213,8 @@
// in an appsink so we can pull the data from each
// channel. Pipeline looks like:
// ... deinterleave ! queue ! appsink.
- GstElement* queue = gst_element_factory_make("queue", 0);
- GstElement* sink = gst_element_factory_make("appsink", 0);
+ GstElement* queue = gst_element_factory_make("queue", nullptr);
+ GstElement* sink = gst_element_factory_make("appsink", nullptr);
static GstAppSinkCallbacks callbacks = {
nullptr, // eos
@@ -225,9 +225,9 @@
},
{ nullptr }
};
- gst_app_sink_set_callbacks(GST_APP_SINK(sink), &callbacks, this, 0);
+ gst_app_sink_set_callbacks(GST_APP_SINK(sink), &callbacks, this, nullptr);
- g_object_set(sink, "sync", FALSE, NULL);
+ g_object_set(sink, "sync", FALSE, nullptr);
gst_bin_add_many(GST_BIN(m_pipeline.get()), queue, sink, nullptr);
@@ -256,12 +256,12 @@
// A decodebin pad was added, plug in a deinterleave element to
// separate each planar channel. Sub pipeline looks like
// ... decodebin2 ! audioconvert ! audioresample ! capsfilter ! deinterleave.
- GstElement* audioConvert = gst_element_factory_make("audioconvert", 0);
- GstElement* audioResample = gst_element_factory_make("audioresample", 0);
- GstElement* capsFilter = gst_element_factory_make("capsfilter", 0);
+ GstElement* audioConvert = gst_element_factory_make("audioconvert", nullptr);
+ GstElement* audioResample = gst_element_factory_make("audioresample", nullptr);
+ GstElement* capsFilter = gst_element_factory_make("capsfilter", nullptr);
m_deInterleave = gst_element_factory_make("deinterleave", "deinterleave");
- g_object_set(m_deInterleave.get(), "keep-positions", TRUE, NULL);
+ g_object_set(m_deInterleave.get(), "keep-positions", TRUE, nullptr);
g_signal_connect_swapped(m_deInterleave.get(), "pad-added", G_CALLBACK(deinterleavePadAddedCallback), this);
g_signal_connect_swapped(m_deInterleave.get(), "no-more-pads", G_CALLBACK(deinterleaveReadyCallback), this);
@@ -316,18 +316,18 @@
GstElement* source;
if (m_data) {
ASSERT(m_dataSize);
- source = gst_element_factory_make("giostreamsrc", 0);
- GRefPtr<GInputStream> memoryStream = adoptGRef(g_memory_input_stream_new_from_data(m_data, m_dataSize, 0));
- g_object_set(source, "stream", memoryStream.get(), NULL);
+ source = gst_element_factory_make("giostreamsrc", nullptr);
+ GRefPtr<GInputStream> memoryStream = adoptGRef(g_memory_input_stream_new_from_data(m_data, m_dataSize, nullptr));
+ g_object_set(source, "stream", memoryStream.get(), nullptr);
} else {
- source = gst_element_factory_make("filesrc", 0);
- g_object_set(source, "location", m_filePath, NULL);
+ source = gst_element_factory_make("filesrc", nullptr);
+ g_object_set(source, "location", m_filePath, nullptr);
}
m_decodebin = gst_element_factory_make("decodebin", "decodebin");
g_signal_connect_swapped(m_decodebin.get(), "pad-added", G_CALLBACK(decodebinPadAddedCallback), this);
- gst_bin_add_many(GST_BIN(m_pipeline.get()), source, m_decodebin.get(), NULL);
+ gst_bin_add_many(GST_BIN(m_pipeline.get()), source, m_decodebin.get(), nullptr);
gst_element_link_pads_full(source, "src", m_decodebin.get(), "sink", GST_PAD_LINK_CHECK_NOTHING);
// Catch errors here immediately, there might not be an error message if we're unlucky.
Modified: releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp (213812 => 213813)
--- releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp 2017-03-13 10:40:05 UTC (rev 213812)
+++ releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp 2017-03-13 10:41:21 UTC (rev 213813)
@@ -83,7 +83,7 @@
}
AudioSourceProviderGStreamer::AudioSourceProviderGStreamer()
- : m_client(0)
+ : m_client(nullptr)
, m_deinterleaveSourcePads(0)
, m_deinterleavePadAddedHandlerId(0)
, m_deinterleaveNoMorePadsHandlerId(0)
@@ -113,13 +113,13 @@
m_audioSinkBin = audioBin;
GstElement* audioTee = gst_element_factory_make("tee", "audioTee");
- GstElement* audioQueue = gst_element_factory_make("queue", 0);
- GstElement* audioConvert = gst_element_factory_make("audioconvert", 0);
- GstElement* audioConvert2 = gst_element_factory_make("audioconvert", 0);
- GstElement* audioResample = gst_element_factory_make("audioresample", 0);
- GstElement* audioResample2 = gst_element_factory_make("audioresample", 0);
+ GstElement* audioQueue = gst_element_factory_make("queue", nullptr);
+ GstElement* audioConvert = gst_element_factory_make("audioconvert", nullptr);
+ GstElement* audioConvert2 = gst_element_factory_make("audioconvert", nullptr);
+ GstElement* audioResample = gst_element_factory_make("audioresample", nullptr);
+ GstElement* audioResample2 = gst_element_factory_make("audioresample", nullptr);
GstElement* volumeElement = gst_element_factory_make("volume", "volume");
- GstElement* audioSink = gst_element_factory_make("autoaudiosink", 0);
+ GstElement* audioSink = gst_element_factory_make("autoaudiosink", nullptr);
gst_bin_add_many(GST_BIN(m_audioSinkBin.get()), audioTee, audioQueue, audioConvert, audioResample, volumeElement, audioConvert2, audioResample2, audioSink, nullptr);
@@ -211,10 +211,10 @@
// The audioconvert and audioresample elements are needed to
// ensure deinterleave and the sinks downstream receive buffers in
// the format specified by the capsfilter.
- GstElement* audioQueue = gst_element_factory_make("queue", 0);
- GstElement* audioConvert = gst_element_factory_make("audioconvert", 0);
- GstElement* audioResample = gst_element_factory_make("audioresample", 0);
- GstElement* capsFilter = gst_element_factory_make("capsfilter", 0);
+ GstElement* audioQueue = gst_element_factory_make("queue", nullptr);
+ GstElement* audioConvert = gst_element_factory_make("audioconvert", nullptr);
+ GstElement* audioResample = gst_element_factory_make("audioresample", nullptr);
+ GstElement* capsFilter = gst_element_factory_make("capsfilter", nullptr);
GstElement* deInterleave = gst_element_factory_make("deinterleave", "deinterleave");
g_object_set(deInterleave, "keep-positions", TRUE, nullptr);
@@ -257,8 +257,8 @@
if (m_deinterleaveSourcePads > 2) {
g_warning("The AudioSourceProvider supports only mono and stereo audio. Silencing out this new channel.");
- GstElement* queue = gst_element_factory_make("queue", 0);
- GstElement* sink = gst_element_factory_make("fakesink", 0);
+ GstElement* queue = gst_element_factory_make("queue", nullptr);
+ GstElement* sink = gst_element_factory_make("fakesink", nullptr);
g_object_set(sink, "async", FALSE, nullptr);
gst_bin_add_many(GST_BIN(m_audioSinkBin.get()), queue, sink, nullptr);
@@ -277,14 +277,14 @@
// in an appsink so we can pull the data from each
// channel. Pipeline looks like:
// ... deinterleave ! queue ! appsink.
- GstElement* queue = gst_element_factory_make("queue", 0);
- GstElement* sink = gst_element_factory_make("appsink", 0);
+ GstElement* queue = gst_element_factory_make("queue", nullptr);
+ GstElement* sink = gst_element_factory_make("appsink", nullptr);
GstAppSinkCallbacks callbacks;
- callbacks.eos = 0;
- callbacks.new_preroll = 0;
+ callbacks.eos = nullptr;
+ callbacks.new_preroll = nullptr;
callbacks.new_sample = onAppsinkNewBufferCallback;
- gst_app_sink_set_callbacks(GST_APP_SINK(sink), &callbacks, this, 0);
+ gst_app_sink_set_callbacks(GST_APP_SINK(sink), &callbacks, this, nullptr);
g_object_set(sink, "async", FALSE, nullptr);
Modified: releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp (213812 => 213813)
--- releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp 2017-03-13 10:40:05 UTC (rev 213812)
+++ releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp 2017-03-13 10:41:21 UTC (rev 213813)
@@ -184,14 +184,14 @@
src->priv = priv;
new (priv) WebKitWebAudioSourcePrivate();
- priv->sourcePad = webkitGstGhostPadFromStaticTemplate(&srcTemplate, "src", 0);
+ priv->sourcePad = webkitGstGhostPadFromStaticTemplate(&srcTemplate, "src", nullptr);
gst_element_add_pad(GST_ELEMENT(src), priv->sourcePad);
- priv->provider = 0;
- priv->bus = 0;
+ priv->provider = nullptr;
+ priv->bus = nullptr;
g_rec_mutex_init(&priv->mutex);
- priv->task = adoptGRef(gst_task_new(reinterpret_cast<GstTaskFunction>(webKitWebAudioSrcLoop), src, 0));
+ priv->task = adoptGRef(gst_task_new(reinterpret_cast<GstTaskFunction>(webKitWebAudioSrcLoop), src, nullptr));
gst_task_set_lock(priv->task.get(), &priv->mutex);
}
@@ -344,7 +344,7 @@
}
// FIXME: Add support for local/live audio input.
- priv->provider->render(0, priv->bus, priv->framesToPull);
+ priv->provider->render(nullptr, priv->bus, priv->framesToPull);
ASSERT(channelBufferList.size() == priv->sources.size());
bool failed = false;
@@ -378,7 +378,7 @@
case GST_STATE_CHANGE_NULL_TO_READY:
if (!src->priv->interleave) {
gst_element_post_message(element, gst_missing_element_message_new(element, "interleave"));
- GST_ELEMENT_ERROR(src, CORE, MISSING_PLUGIN, (0), ("no interleave"));
+ GST_ELEMENT_ERROR(src, CORE, MISSING_PLUGIN, (nullptr), ("no interleave"));
return GST_STATE_CHANGE_FAILURE;
}
src->priv->numberOfSamples = 0;
Modified: releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/AudioTrackPrivateGStreamer.cpp (213812 => 213813)
--- releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/AudioTrackPrivateGStreamer.cpp 2017-03-13 10:40:05 UTC (rev 213812)
+++ releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/AudioTrackPrivateGStreamer.cpp 2017-03-13 10:41:21 UTC (rev 213813)
@@ -55,7 +55,7 @@
AudioTrackPrivate::setEnabled(enabled);
if (enabled && m_playbin)
- g_object_set(m_playbin.get(), "current-audio", m_index, NULL);
+ g_object_set(m_playbin.get(), "current-audio", m_index, nullptr);
}
} // namespace WebCore
Modified: releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/GStreamerUtilities.cpp (213812 => 213813)
--- releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/GStreamerUtilities.cpp 2017-03-13 10:40:05 UTC (rev 213812)
+++ releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/GStreamerUtilities.cpp 2017-03-13 10:41:21 UTC (rev 213813)
@@ -152,7 +152,7 @@
GUniqueOutPtr<GError> error;
// FIXME: We should probably pass the arguments from the command line.
- bool gstInitialized = gst_init_check(0, 0, &error.outPtr());
+ bool gstInitialized = gst_init_check(nullptr, nullptr, &error.outPtr());
ASSERT_WITH_MESSAGE(gstInitialized, "GStreamer initialization failed: %s", error ? error->message : "unknown error occurred");
#if ENABLE(VIDEO_TRACK) && USE(GSTREAMER_MPEGTS)
Modified: releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp (213812 => 213813)
--- releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp 2017-03-13 10:40:05 UTC (rev 213812)
+++ releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp 2017-03-13 10:41:21 UTC (rev 213813)
@@ -96,8 +96,8 @@
return;
const char* role = m_player->client().mediaPlayerIsVideo() ? "video" : "music";
- GstStructure* structure = gst_structure_new("stream-properties", "media.role", G_TYPE_STRING, role, NULL);
- g_object_set(object, "stream-properties", structure, NULL);
+ GstStructure* structure = gst_structure_new("stream-properties", "media.role", G_TYPE_STRING, role, nullptr);
+ g_object_set(object, "stream-properties", structure, nullptr);
gst_structure_free(structure);
GUniquePtr<gchar> elementName(gst_element_get_name(GST_ELEMENT(object)));
GST_DEBUG("Set media.role as %s at %s", role, elementName.get());
@@ -107,7 +107,7 @@
{
if (isAvailable())
registrar([](MediaPlayer* player) { return std::make_unique<MediaPlayerPrivateGStreamer>(player); },
- getSupportedTypes, supportsType, 0, 0, 0, supportsKeySystem);
+ getSupportedTypes, supportsType, nullptr, nullptr, nullptr, supportsKeySystem);
}
bool initializeGStreamerAndRegisterWebKitElements()
@@ -120,7 +120,7 @@
GRefPtr<GstElementFactory> srcFactory = adoptGRef(gst_element_factory_find("webkitwebsrc"));
if (!srcFactory) {
GST_DEBUG_CATEGORY_INIT(webkit_media_player_debug, "webkitmediaplayer", 0, "WebKit media player");
- gst_element_register(0, "webkitwebsrc", GST_RANK_PRIMARY + 100, WEBKIT_TYPE_WEB_SRC);
+ gst_element_register(nullptr, "webkitwebsrc", GST_RANK_PRIMARY + 100, WEBKIT_TYPE_WEB_SRC);
}
return true;
@@ -153,10 +153,10 @@
, m_seeking(false)
, m_seekIsPending(false)
, m_seekTime(0)
- , m_source(0)
+ , m_source(nullptr)
, m_volumeAndMuteInitialized(false)
, m_weakPtrFactory(this)
- , m_mediaLocations(0)
+ , m_mediaLocations(nullptr)
, m_mediaLocationCurrentIndex(0)
, m_playbackRatePause(false)
, m_timeOfOverlappingSeek(-1)
@@ -194,7 +194,7 @@
if (m_mediaLocations) {
gst_structure_free(m_mediaLocations);
- m_mediaLocations = 0;
+ m_mediaLocations = nullptr;
}
if (WEBKIT_IS_WEB_SRC(m_source.get()) && GST_OBJECT_PARENT(m_source.get()))
@@ -796,7 +796,7 @@
gst_pad_get_sticky_event(m_textAppSinkPad.get(), GST_EVENT_STREAM_START, 0));
GRefPtr<GstSample> sample;
- g_signal_emit_by_name(m_textAppSink.get(), "pull-sample", &sample.outPtr(), NULL);
+ g_signal_emit_by_name(m_textAppSink.get(), "pull-sample", &sample.outPtr(), nullptr);
ASSERT(sample);
if (streamStartEvent) {
@@ -986,7 +986,7 @@
// Construct a filename for the graphviz dot file output.
GstState newState;
- gst_message_parse_state_changed(message, ¤tState, &newState, 0);
+ gst_message_parse_state_changed(message, ¤tState, &newState, nullptr);
CString dotFileName = String::format("webkit-video.%s_%s", gst_element_state_get_name(currentState), gst_element_state_get_name(newState)).utf8();
GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS(GST_BIN(m_pipeline.get()), GST_DEBUG_GRAPH_SHOW_ALL, dotFileName.data());
@@ -1173,12 +1173,11 @@
ASSERT(toc);
for (GList* i = gst_toc_get_entries(toc.get()); i; i = i->next)
- processTableOfContentsEntry(static_cast<GstTocEntry*>(i->data), 0);
+ processTableOfContentsEntry(static_cast<GstTocEntry*>(i->data));
}
-void MediaPlayerPrivateGStreamer::processTableOfContentsEntry(GstTocEntry* entry, GstTocEntry* parent)
+void MediaPlayerPrivateGStreamer::processTableOfContentsEntry(GstTocEntry* entry)
{
- UNUSED_PARAM(parent);
ASSERT(entry);
RefPtr<GenericCueData> cue = GenericCueData::create();
@@ -1192,7 +1191,7 @@
GstTagList* tags = gst_toc_entry_get_tags(entry);
if (tags) {
- gchar* title = 0;
+ gchar* title = nullptr;
gst_tag_list_get_string(tags, GST_TAG_TITLE, &title);
if (title) {
cue->setContent(title);
@@ -1203,7 +1202,7 @@
m_chaptersTrack->addGenericCue(cue.release());
for (GList* i = gst_toc_entry_get_sub_entries(entry); i; i = i->next)
- processTableOfContentsEntry(static_cast<GstTocEntry*>(i->data), entry);
+ processTableOfContentsEntry(static_cast<GstTocEntry*>(i->data));
}
#endif
@@ -1219,7 +1218,7 @@
gint64 start, stop;
gdouble fillStatus = 100.0;
- gst_query_parse_buffering_range(query, 0, &start, &stop, 0);
+ gst_query_parse_buffering_range(query, nullptr, &start, &stop, nullptr);
gst_query_unref(query);
if (stop != -1)
@@ -1654,7 +1653,7 @@
return false;
const GValue* locations = gst_structure_get_value(m_mediaLocations, "locations");
- const gchar* newLocation = 0;
+ const gchar* newLocation = nullptr;
if (!locations) {
// Fallback on new-location string.
@@ -1665,7 +1664,7 @@
if (!newLocation) {
if (m_mediaLocationCurrentIndex < 0) {
- m_mediaLocations = 0;
+ m_mediaLocations = nullptr;
return false;
}
@@ -1974,7 +1973,7 @@
GstElement* MediaPlayerPrivateGStreamer::createAudioSink()
{
- m_autoAudioSink = gst_element_factory_make("autoaudiosink", 0);
+ m_autoAudioSink = gst_element_factory_make("autoaudiosink", nullptr);
if (!m_autoAudioSink) {
GST_WARNING("GStreamer's autoaudiosink not found. Please check your gst-plugins-good installation");
return nullptr;
@@ -2115,7 +2114,7 @@
// See https://bugzilla.gnome.org/show_bug.cgi?id=735748 for
// the reason for using >= 1.4.2 instead of >= 1.4.0.
if (m_preservesPitch && webkitGstCheckVersion(1, 4, 2)) {
- GstElement* scale = gst_element_factory_make("scaletempo", 0);
+ GstElement* scale = gst_element_factory_make("scaletempo", nullptr);
if (!scale)
GST_WARNING("Failed to create scaletempo");
Modified: releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.h (213812 => 213813)
--- releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.h 2017-03-13 10:40:05 UTC (rev 213812)
+++ releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.h 2017-03-13 10:41:21 UTC (rev 213813)
@@ -161,7 +161,7 @@
#endif
#if ENABLE(VIDEO_TRACK)
void processTableOfContents(GstMessage*);
- void processTableOfContentsEntry(GstTocEntry*, GstTocEntry* parent);
+ void processTableOfContentsEntry(GstTocEntry*);
#endif
virtual bool doSeek(gint64 position, float rate, GstSeekFlags seekType);
virtual void updatePlaybackRate();
Modified: releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamerBase.cpp (213812 => 213813)
--- releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamerBase.cpp 2017-03-13 10:40:05 UTC (rev 213812)
+++ releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamerBase.cpp 2017-03-13 10:41:21 UTC (rev 213813)
@@ -201,7 +201,7 @@
MediaPlayerPrivateGStreamerBase::MediaPlayerPrivateGStreamerBase(MediaPlayer* player)
: m_player(player)
- , m_fpsSink(0)
+ , m_fpsSink(nullptr)
, m_readyState(MediaPlayer::HaveNothing)
, m_networkState(MediaPlayer::Empty)
#if USE(GSTREAMER_GL) || USE(COORDINATED_GRAPHICS_THREADED)
@@ -209,7 +209,7 @@
#endif
, m_usingFallbackVideoSink(false)
#if ENABLE(LEGACY_ENCRYPTED_MEDIA)
- , m_cdmSession(0)
+ , m_cdmSession(nullptr)
#endif
{
g_mutex_init(&m_sampleMutex);
@@ -595,7 +595,7 @@
if (!m_volumeElement)
return;
- g_object_set(m_volumeElement.get(), "mute", muted, NULL);
+ g_object_set(m_volumeElement.get(), "mute", muted, nullptr);
}
bool MediaPlayerPrivateGStreamerBase::muted() const
@@ -604,7 +604,7 @@
return false;
bool muted;
- g_object_get(m_volumeElement.get(), "mute", &muted, NULL);
+ g_object_get(m_volumeElement.get(), "mute", &muted, nullptr);
return muted;
}
@@ -614,7 +614,7 @@
return;
gboolean muted;
- g_object_get(m_volumeElement.get(), "mute", &muted, NULL);
+ g_object_get(m_volumeElement.get(), "mute", &muted, nullptr);
m_player->muteChanged(static_cast<bool>(muted));
}
@@ -1179,12 +1179,12 @@
// https://bugs.webkit.org/show_bug.cgi?id=118974 for more information.
if (!m_player->platformVolumeConfigurationRequired()) {
GST_DEBUG("Setting stream volume to %f", m_player->volume());
- g_object_set(m_volumeElement.get(), "volume", m_player->volume(), NULL);
+ g_object_set(m_volumeElement.get(), "volume", m_player->volume(), nullptr);
} else
GST_DEBUG("Not setting stream volume, trusting system one");
GST_DEBUG("Setting stream muted %d", m_player->muted());
- g_object_set(m_volumeElement.get(), "mute", m_player->muted(), NULL);
+ g_object_set(m_volumeElement.get(), "mute", m_player->muted(), nullptr);
g_signal_connect_swapped(m_volumeElement.get(), "notify::volume", G_CALLBACK(volumeChangedCallback), this);
g_signal_connect_swapped(m_volumeElement.get(), "notify::mute", G_CALLBACK(muteChangedCallback), this);
@@ -1194,7 +1194,7 @@
{
guint64 decodedFrames = 0;
if (m_fpsSink)
- g_object_get(m_fpsSink.get(), "frames-rendered", &decodedFrames, NULL);
+ g_object_get(m_fpsSink.get(), "frames-rendered", &decodedFrames, nullptr);
return static_cast<unsigned>(decodedFrames);
}
@@ -1202,7 +1202,7 @@
{
guint64 framesDropped = 0;
if (m_fpsSink)
- g_object_get(m_fpsSink.get(), "frames-dropped", &framesDropped, NULL);
+ g_object_get(m_fpsSink.get(), "frames-dropped", &framesDropped, nullptr);
return static_cast<unsigned>(framesDropped);
}
Modified: releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamerOwr.cpp (213812 => 213813)
--- releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamerOwr.cpp 2017-03-13 10:40:05 UTC (rev 213812)
+++ releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamerOwr.cpp 2017-03-13 10:41:21 UTC (rev 213813)
@@ -286,7 +286,7 @@
if (initializeGStreamerAndGStreamerDebugging()) {
registrar([](MediaPlayer* player) {
return std::make_unique<MediaPlayerPrivateGStreamerOwr>(player);
- }, getSupportedTypes, supportsType, 0, 0, 0, 0);
+ }, getSupportedTypes, supportsType, nullptr, nullptr, nullptr, nullptr);
}
}
Modified: releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/TextCombinerGStreamer.cpp (213812 => 213813)
--- releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/TextCombinerGStreamer.cpp 2017-03-13 10:40:05 UTC (rev 213812)
+++ releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/TextCombinerGStreamer.cpp 2017-03-13 10:41:21 UTC (rev 213813)
@@ -76,7 +76,7 @@
static void webkit_text_combiner_init(WebKitTextCombiner* combiner)
{
- combiner->funnel = gst_element_factory_make("funnel", NULL);
+ combiner->funnel = gst_element_factory_make("funnel", nullptr);
ASSERT(combiner->funnel);
gboolean ret = gst_bin_add(GST_BIN(combiner), combiner->funnel);
@@ -147,7 +147,7 @@
* the funnel */
if (targetParent == combiner->funnel) {
/* Setup a WebVTT encoder */
- GstElement* encoder = gst_element_factory_make("webvttenc", NULL);
+ GstElement* encoder = gst_element_factory_make("webvttenc", nullptr);
ASSERT(encoder);
ret = gst_bin_add(GST_BIN(combiner), encoder);
@@ -232,7 +232,7 @@
GstPad* pad = gst_element_request_pad(combiner->funnel, templ, name, caps);
ASSERT(pad);
- GstPad* ghostPad = GST_PAD(g_object_new(WEBKIT_TYPE_TEXT_COMBINER_PAD, "direction", gst_pad_get_direction(pad), NULL));
+ GstPad* ghostPad = GST_PAD(g_object_new(WEBKIT_TYPE_TEXT_COMBINER_PAD, "direction", gst_pad_get_direction(pad), nullptr));
ASSERT(ghostPad);
ret = gst_ghost_pad_construct(GST_GHOST_PAD(ghostPad));
@@ -295,7 +295,7 @@
GstElement* webkitTextCombinerNew()
{
- return GST_ELEMENT(g_object_new(WEBKIT_TYPE_TEXT_COMBINER, 0));
+ return GST_ELEMENT(g_object_new(WEBKIT_TYPE_TEXT_COMBINER, nullptr));
}
#endif // ENABLE(VIDEO) && USE(GSTREAMER) && ENABLE(VIDEO_TRACK)
Modified: releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/TextSinkGStreamer.cpp (213812 => 213813)
--- releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/TextSinkGStreamer.cpp 2017-03-13 10:40:05 UTC (rev 213812)
+++ releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/TextSinkGStreamer.cpp 2017-03-13 10:41:21 UTC (rev 213813)
@@ -95,7 +95,7 @@
GstElement* webkitTextSinkNew()
{
- return GST_ELEMENT(g_object_new(WEBKIT_TYPE_TEXT_SINK, 0));
+ return GST_ELEMENT(g_object_new(WEBKIT_TYPE_TEXT_SINK, nullptr));
}
#endif // ENABLE(VIDEO) && USE(GSTREAMER) && ENABLE(VIDEO_TRACK)
Modified: releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/TrackPrivateBaseGStreamer.cpp (213812 => 213813)
--- releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/TrackPrivateBaseGStreamer.cpp 2017-03-13 10:40:05 UTC (rev 213812)
+++ releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/TrackPrivateBaseGStreamer.cpp 2017-03-13 10:41:21 UTC (rev 213813)
@@ -89,7 +89,7 @@
{
GRefPtr<GstTagList> tags;
if (g_object_class_find_property(G_OBJECT_GET_CLASS(m_pad.get()), "tags"))
- g_object_get(m_pad.get(), "tags", &tags.outPtr(), NULL);
+ g_object_get(m_pad.get(), "tags", &tags.outPtr(), nullptr);
else
tags = adoptGRef(gst_tag_list_new_empty());
@@ -108,7 +108,7 @@
gboolean active = false;
if (m_pad && g_object_class_find_property(G_OBJECT_GET_CLASS(m_pad.get()), "active"))
- g_object_get(m_pad.get(), "active", &active, NULL);
+ g_object_get(m_pad.get(), "active", &active, nullptr);
setActive(active);
}
Modified: releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/VideoSinkGStreamer.cpp (213812 => 213813)
--- releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/VideoSinkGStreamer.cpp 2017-03-13 10:40:05 UTC (rev 213812)
+++ releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/VideoSinkGStreamer.cpp 2017-03-13 10:41:21 UTC (rev 213813)
@@ -183,7 +183,7 @@
static void webkit_video_sink_init(WebKitVideoSink* sink)
{
sink->priv = G_TYPE_INSTANCE_GET_PRIVATE(sink, WEBKIT_TYPE_VIDEO_SINK, WebKitVideoSinkPrivate);
- g_object_set(GST_BASE_SINK(sink), "enable-last-sample", FALSE, NULL);
+ g_object_set(GST_BASE_SINK(sink), "enable-last-sample", FALSE, nullptr);
new (sink->priv) WebKitVideoSinkPrivate();
}
@@ -341,7 +341,7 @@
static gboolean webkitVideoSinkProposeAllocation(GstBaseSink* baseSink, GstQuery* query)
{
GstCaps* caps;
- gst_query_parse_allocation(query, &caps, 0);
+ gst_query_parse_allocation(query, &caps, nullptr);
if (!caps)
return FALSE;
@@ -349,9 +349,9 @@
if (!gst_video_info_from_caps(&sink->priv->info, caps))
return FALSE;
- gst_query_add_allocation_meta(query, GST_VIDEO_META_API_TYPE, 0);
- gst_query_add_allocation_meta(query, GST_VIDEO_CROP_META_API_TYPE, 0);
- gst_query_add_allocation_meta(query, GST_VIDEO_GL_TEXTURE_UPLOAD_META_API_TYPE, 0);
+ gst_query_add_allocation_meta(query, GST_VIDEO_META_API_TYPE, nullptr);
+ gst_query_add_allocation_meta(query, GST_VIDEO_CROP_META_API_TYPE, nullptr);
+ gst_query_add_allocation_meta(query, GST_VIDEO_GL_TEXTURE_UPLOAD_META_API_TYPE, nullptr);
return TRUE;
}
@@ -408,7 +408,7 @@
GstElement* webkitVideoSinkNew()
{
- return GST_ELEMENT(g_object_new(WEBKIT_TYPE_VIDEO_SINK, 0));
+ return GST_ELEMENT(g_object_new(WEBKIT_TYPE_VIDEO_SINK, nullptr));
}
#endif // ENABLE(VIDEO) && USE(GSTREAMER)
Modified: releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/VideoTrackPrivateGStreamer.cpp (213812 => 213813)
--- releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/VideoTrackPrivateGStreamer.cpp 2017-03-13 10:40:05 UTC (rev 213812)
+++ releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/VideoTrackPrivateGStreamer.cpp 2017-03-13 10:41:21 UTC (rev 213813)
@@ -55,7 +55,7 @@
VideoTrackPrivate::setSelected(selected);
if (selected && m_playbin)
- g_object_set(m_playbin.get(), "current-video", m_index, NULL);
+ g_object_set(m_playbin.get(), "current-video", m_index, nullptr);
}
} // namespace WebCore
Modified: releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/WebKitWebSourceGStreamer.cpp (213812 => 213813)
--- releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/WebKitWebSourceGStreamer.cpp 2017-03-13 10:40:05 UTC (rev 213812)
+++ releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/WebKitWebSourceGStreamer.cpp 2017-03-13 10:41:21 UTC (rev 213813)
@@ -270,7 +270,7 @@
priv->createdInMainThread = isMainThread();
- priv->appsrc = GST_APP_SRC(gst_element_factory_make("appsrc", 0));
+ priv->appsrc = GST_APP_SRC(gst_element_factory_make("appsrc", nullptr));
if (!priv->appsrc) {
GST_ERROR_OBJECT(src, "Failed to create appsrc");
return;
@@ -287,7 +287,7 @@
GST_OBJECT_FLAG_SET(priv->srcpad, GST_PAD_FLAG_NEED_PARENT);
gst_pad_set_query_function(priv->srcpad, webKitWebSrcQueryWithParent);
- gst_app_src_set_callbacks(priv->appsrc, &appsrcCallbacks, src, 0);
+ gst_app_src_set_callbacks(priv->appsrc, &appsrcCallbacks, src, nullptr);
gst_app_src_set_emit_signals(priv->appsrc, FALSE);
gst_app_src_set_stream_type(priv->appsrc, GST_APP_STREAM_TYPE_SEEKABLE);
@@ -308,9 +308,9 @@
// likely that libsoup already provides new data before
// the queue is really empty.
// This might need tweaking for ports not using libsoup.
- g_object_set(priv->appsrc, "min-percent", 20, NULL);
+ g_object_set(priv->appsrc, "min-percent", 20, nullptr);
- gst_app_src_set_caps(priv->appsrc, 0);
+ gst_app_src_set_caps(priv->appsrc, nullptr);
gst_app_src_set_size(priv->appsrc, -1);
}
@@ -319,7 +319,7 @@
WebKitWebSrc* src = ""
WebKitWebSrcPrivate* priv = src->priv;
- priv->player = 0;
+ priv->player = nullptr;
GST_CALL_PARENT(G_OBJECT_CLASS, dispose, (object));
}
@@ -339,7 +339,7 @@
switch (propID) {
case PROP_LOCATION:
- gst_uri_handler_set_uri(reinterpret_cast<GstURIHandler*>(src), g_value_get_string(value), 0);
+ gst_uri_handler_set_uri(reinterpret_cast<GstURIHandler*>(src), g_value_get_string(value), nullptr);
break;
case PROP_KEEP_ALIVE:
src->priv->keepAlive = g_value_get_boolean(value);
@@ -433,13 +433,13 @@
if (!wasSeeking) {
priv->size = 0;
priv->requestedOffset = 0;
- priv->player = 0;
+ priv->player = nullptr;
}
locker.unlock();
if (priv->appsrc) {
- gst_app_src_set_caps(priv->appsrc, 0);
+ gst_app_src_set_caps(priv->appsrc, nullptr);
if (!wasSeeking)
gst_app_src_set_size(priv->appsrc, -1);
}
@@ -615,7 +615,7 @@
if (!priv->appsrc) {
gst_element_post_message(element,
gst_missing_element_message_new(element, "appsrc"));
- GST_ELEMENT_ERROR(src, CORE, MISSING_PLUGIN, (0), ("no appsrc"));
+ GST_ELEMENT_ERROR(src, CORE, MISSING_PLUGIN, (nullptr), ("no appsrc"));
return GST_STATE_CHANGE_FAILURE;
}
break;
@@ -658,7 +658,7 @@
case GST_QUERY_DURATION: {
GstFormat format;
- gst_query_parse_duration(query, &format, NULL);
+ gst_query_parse_duration(query, &format, nullptr);
GST_DEBUG_OBJECT(src, "duration query in format %s", gst_format_get_name(format));
WTF::GMutexLocker<GMutex> locker(*GST_OBJECT_GET_LOCK(src));
@@ -710,7 +710,7 @@
const gchar* const* webKitWebSrcGetProtocols(GType)
{
- static const char* protocols[] = {"http", "https", "blob", 0 };
+ static const char* protocols[] = {"http", "https", "blob", nullptr };
return protocols;
}
@@ -936,7 +936,7 @@
} else
gst_app_src_set_size(priv->appsrc, -1);
- gst_app_src_set_caps(priv->appsrc, 0);
+ gst_app_src_set_caps(priv->appsrc, nullptr);
}
void StreamingClient::handleDataReceived(const char* data, int length)
@@ -1003,7 +1003,7 @@
GstFlowReturn ret = gst_app_src_push_buffer(priv->appsrc, priv->buffer.leakRef());
if (ret != GST_FLOW_OK && ret != GST_FLOW_EOS)
- GST_ELEMENT_ERROR(src, CORE, FAILED, (0), (0));
+ GST_ELEMENT_ERROR(src, CORE, FAILED, (nullptr), (nullptr));
}
void StreamingClient::handleNotifyFinished()
@@ -1182,7 +1182,7 @@
WebKitWebSrc* src = ""
GST_ERROR_OBJECT(src, "Have failure: %s", error.localizedDescription().utf8().data());
- GST_ELEMENT_ERROR(src, RESOURCE, FAILED, ("%s", error.localizedDescription().utf8().data()), (0));
+ GST_ELEMENT_ERROR(src, RESOURCE, FAILED, ("%s", error.localizedDescription().utf8().data()), (nullptr));
gst_app_src_end_of_stream(src->priv->appsrc);
}
@@ -1197,7 +1197,7 @@
uri.reset(g_strdup(src->priv->originalURI.data()));
locker.unlock();
- GST_ELEMENT_ERROR(src, RESOURCE, OPEN_READ, ("Access to \"%s\" was blocked", uri.get()), (0));
+ GST_ELEMENT_ERROR(src, RESOURCE, OPEN_READ, ("Access to \"%s\" was blocked", uri.get()), (nullptr));
}
void ResourceHandleStreamingClient::cannotShowURL(ResourceHandle*)
@@ -1211,7 +1211,7 @@
uri.reset(g_strdup(src->priv->originalURI.data()));
locker.unlock();
- GST_ELEMENT_ERROR(src, RESOURCE, OPEN_READ, ("Can't show \"%s\"", uri.get()), (0));
+ GST_ELEMENT_ERROR(src, RESOURCE, OPEN_READ, ("Can't show \"%s\"", uri.get()), (nullptr));
}
#endif // USE(GSTREAMER)
Modified: releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/mse/WebKitMediaSourceGStreamer.cpp (213812 => 213813)
--- releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/mse/WebKitMediaSourceGStreamer.cpp 2017-03-13 10:40:05 UTC (rev 213812)
+++ releases/WebKitGTK/webkit-2.16/Source/WebCore/platform/graphics/gstreamer/mse/WebKitMediaSourceGStreamer.cpp 2017-03-13 10:41:21 UTC (rev 213813)
@@ -593,7 +593,7 @@
const gchar* const* webKitMediaSrcGetProtocols(GType)
{
- static const char* protocols[] = {"mediasourceblob", 0 };
+ static const char* protocols[] = {"mediasourceblob", nullptr };
return protocols;
}
