[Webkit-unassigned] [Bug 105292] [GStreamer] GstMessage handler in AudioDestination

[bugzilla-daemon at webkit.org]

https://bugs.webkit.org/show_bug.cgi?id=105292





--- Comment #2 from Martin Robinson <mrobinson at webkit.org>  2012-12-27 10:14:03 PST ---
(From update of attachment 179941)
View in context: https://bugs.webkit.org/attachment.cgi?id=179941&action=review

> Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.cpp:68
> +    GRefPtr<GstBus> bus = gst_pipeline_get_bus(GST_PIPELINE(m_pipeline));
> +    gst_bus_add_signal_watch(bus.get());
> +    g_signal_connect(bus.get(), "message", G_CALLBACK(messageCallback), this);

See below.

> Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.cpp:91
> +    GRefPtr<GstBus> bus = gst_pipeline_get_bus(GST_PIPELINE(m_pipeline));
> +    g_signal_handlers_disconnect_by_func(bus.get(), reinterpret_cast<gpointer>(messageCallback), this);

Why not just use a raw pointer? By using GRefPtr you are doing a ref and unref.

> Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.cpp:140
> +        g_warning("Warning: %d, %s. Debug output: %s", error->code,  error->message, debug.get());

Wouldn't it make sense to use gst_debug_log here?

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