Benjamin Gaignard [Tue, 19 Feb 2013 10:47:20 +0000 (11:47 +0100)]
v4l2: Add support of dmabuf
v4l has add a new IOCTL to export a buffer by using dmabuf.
This patch allow to use this new IOTCL if it has been defined in videodev2.h
I introduce a new IO mode (GST_V4L2_IO_DMABUF) to enable this way of working.
https://bugzilla.gnome.org/show_bug.cgi?id=693826
Tim-Philipp Müller [Mon, 18 Feb 2013 20:04:05 +0000 (20:04 +0000)]
qtdemux: fix up dodgy code that tries to fix up a broken moov atom
After gst_buffer_new_and_alloc() gst_buffer_copy_into() will likely
append to the already-existing memory instead of filling it.
Tim-Philipp Müller [Mon, 18 Feb 2013 16:32:13 +0000 (16:32 +0000)]
qtdemux: fix potential crash on short MOOV atom
Don't unmap short MOOV atom buffer twice, which happened
in the case where we don't fix up the MOOV atom.
Fixes crashes when thumbnailing partial mp4 file where
the MOOV atom is still incomplete.
https://bugzilla.gnome.org/show_bug.cgi?id=694010
Tim-Philipp Müller [Sat, 16 Feb 2013 16:49:22 +0000 (16:49 +0000)]
souphttpsrc: set SOUP_VERSION_{MIN_REQUIRED,MAX_ALLOWED} to suppress deprecations with newer versions
https://bugzilla.gnome.org/show_bug.cgi?id=693911
Tim-Philipp Müller [Sat, 16 Feb 2013 15:47:02 +0000 (15:47 +0000)]
soup: use default proxy resolver instead of deprecated GNOME proxy resolver
Apparently there's no reason to use it any longer. Drop libsoup-gnome
dependency while at it, now that we don't need anything from it any
more (it only consists entirely of deprecated API now anyways).
https://bugzilla.gnome.org/show_bug.cgi?id=693911
Tim-Philipp Müller [Fri, 15 Feb 2013 15:43:43 +0000 (15:43 +0000)]
tests: fix some h264 caps
Doesn't fix anything in particular, but is
still needed here for correctness.
Stefan Sauer [Fri, 15 Feb 2013 07:19:24 +0000 (08:19 +0100)]
audiopanorama: remove channel-mask from caps
The channel-mask is only needed for channels>2 which we don't do.
Benjamin Gaignard [Fri, 15 Feb 2013 15:21:21 +0000 (16:21 +0100)]
v4l2: don't check stride for encoded formats
Don't try to check the stride for encoded formats. Some drivers output
something != 0 and then we don't want to fail on that.
Tim-Philipp Müller [Fri, 15 Feb 2013 14:11:36 +0000 (14:11 +0000)]
udpsrc: use g_socket_set_option() to set buffer size with newer GLib versions
So we have to worry less about portability.
https://bugzilla.gnome.org/show_bug.cgi?id=692400
Tim-Philipp Müller [Thu, 14 Feb 2013 14:13:27 +0000 (14:13 +0000)]
jpegdec: remove sof-marker from template caps for now
Now that the subset check actually works, this breaks
things with demuxers that don't put a "sof-marker"
in their jpeg caps, and we don't have a good parser
to plug either yet.
Sebastian Dröge [Wed, 13 Feb 2013 11:32:10 +0000 (12:32 +0100)]
jpegenc: Put the SOF marker into the caps
Sebastian Dröge [Wed, 13 Feb 2013 11:02:46 +0000 (12:02 +0100)]
rtp-payloading: Fix unit test caps and AMR depayloader sink template caps
Fields were missing from the actual caps, or too many fields
existed in the template caps.
Sebastian Dröge [Wed, 13 Feb 2013 10:53:01 +0000 (11:53 +0100)]
aacparse: Fix caps used in the unit test
The AAC caps passed were incomplete.
Sebastian Dröge [Wed, 13 Feb 2013 10:49:40 +0000 (11:49 +0100)]
wavpack: Fix unit tests, width is now called depth in the caps in 1.0
Tim-Philipp Müller [Tue, 12 Feb 2013 23:31:22 +0000 (23:31 +0000)]
tests: make souphttpsrc unit test work even if http_proxy is set
We're testing with an http server on localhost, but don't support
an exception list for the http_proxy, so just unset the environment
variable to make sure we can run this test properly even if the
environment has http_proxy set.
Also, don't skip all tests if there is an issue with the SSL server,
just run the non-SSL tests then.
https://jenkins.qa.ubuntu.com/view/Raring/view/JHBuild%20Gnome/job/jhbuild-amd64-gst-plugins-good/
Michael Smith [Tue, 12 Feb 2013 20:53:52 +0000 (12:53 -0800)]
qtdemux: extract codec_data for ProRes
Tim 'mithro' Ansell [Thu, 7 Feb 2013 14:02:10 +0000 (01:02 +1100)]
avimux: Fixing buffer leak in gst_avi_mux_do_buffer
gst_avi_mux_do_buffer was leaking data from gst_collect_pads_pop.
Mark Nauwelaerts [Sun, 10 Feb 2013 14:10:32 +0000 (15:10 +0100)]
avidemux: correct duration for audio VBR buffers in pull mode
Mark Nauwelaerts [Fri, 8 Feb 2013 20:28:02 +0000 (21:28 +0100)]
avidemux: proper position reporting and push mode timestamping
... and align current_total semantics in push and pull mode,
which tracks bytes for CBR and blocks for VBR.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=691481
Wim Taymans [Fri, 8 Feb 2013 16:05:27 +0000 (17:05 +0100)]
rtpsession: delay RTCP until first RTP packet
Delay sending the first RTCP packet until we have sent the first RTP packet.
Otherwise we will send out a Receiver Report instead of a sender report.
See https://bugzilla.gnome.org/show_bug.cgi?id=691400
Wim Taymans [Thu, 7 Feb 2013 14:06:40 +0000 (15:06 +0100)]
rtpsession: remove dead code
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=668355
Paul HENRYS [Tue, 29 Jan 2013 09:48:17 +0000 (10:48 +0100)]
rtpptdemux: forward sticky events and then set caps
When a new src pad is added, first forward the sticky events and then
set the caps on the src pad
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692786
Markovtsev Vadim [Thu, 7 Feb 2013 13:32:26 +0000 (14:32 +0100)]
rtpjitterbuffer: improve debug output
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688935
Wim Taymans [Mon, 26 Sep 2011 21:42:51 +0000 (14:42 -0700)]
rtpbin: rework cleanup of streams
Move the work of cleaning up the client streams in the free_stream
function. This allows us to properly clean up the client streams when we
remove an RTP stream as well.
Based on patch by Sujay <sdatar@cisco.com>
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=660156
Tim 'mithro' Ansell [Thu, 7 Feb 2013 10:40:35 +0000 (11:40 +0100)]
videomixer2: avoid caps leak
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=693307
Wim Taymans [Wed, 6 Feb 2013 16:15:11 +0000 (17:15 +0100)]
jitterbuffer: do skew estimation only for new timestamps
Only run the skew estimation code when we have a new RTP timestamp. If we have
the same RTP timestamp, we simply use the previous estimation. This works
because the new observation with the same RTP timestamp has to have a bigger
receiver time and is thus not going to influence the estimation except for
causing more jitter.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=640023
Wim Taymans [Wed, 6 Feb 2013 12:52:26 +0000 (13:52 +0100)]
rtspsrc: only EOS when our source sends BYE
Only EOS when we receive a BYE event from the SSRC of our stream.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=675453
Wim Taymans [Wed, 6 Feb 2013 12:47:51 +0000 (13:47 +0100)]
rtspsrc: save the stream SSRC
Conflicts:
gst/rtsp/gstrtspsrc.c
Wim Taymans [Wed, 6 Feb 2013 12:18:18 +0000 (13:18 +0100)]
rtspsrc: flush connection when stopping
When we stop, we can flush all pending commands so that we can stop and
join the task.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=684924
Stefan Sauer [Tue, 5 Feb 2013 21:02:13 +0000 (22:02 +0100)]
spectrum: remove outdates readme
Lets remove the readme from pre-0.1.0 that is completely irrelevant now.
Stefan Sauer [Tue, 5 Feb 2013 06:32:29 +0000 (07:32 +0100)]
audiopanorama: add more debug logging
Stefan Sauer [Tue, 5 Feb 2013 07:26:14 +0000 (08:26 +0100)]
level-example. avoid taking the arrays again for each channel for clarity
Also introduce some blank lines for better readability and update the comments.
Rico Tzschichholz [Mon, 4 Feb 2013 18:38:41 +0000 (18:38 +0000)]
audioparsers: fix typo in noinst_headers
Stefan Sauer [Mon, 4 Feb 2013 10:08:23 +0000 (11:08 +0100)]
audiopanorama: further port to 1.0
Transformcaps is not called with caps containing single structures anymore. Also add missing filter handling. Still does not negotiate though.
Stefan Sauer [Sun, 3 Feb 2013 21:45:52 +0000 (22:45 +0100)]
audiopanorama: fix caps
We don't turn float into 32bit pcm. Looks like a typo from updating the caps.
Olivier Crête [Sun, 3 Feb 2013 12:14:50 +0000 (13:14 +0100)]
level: Add missing coma between formats
Matthew Waters [Thu, 31 Jan 2013 11:55:18 +0000 (22:55 +1100)]
videomixer: fix eos timestamp check
fixes hang in videotestsrc num-buffers=20 ! videomixer ! fakesink
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692935
Dirk Van Haerenborgh [Thu, 31 Jan 2013 10:35:09 +0000 (11:35 +0100)]
avimux: add support for raw monochrome 8-bit video
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692932
Alexey Chernov [Fri, 18 Jan 2013 17:08:12 +0000 (21:08 +0400)]
osxvideosink: Make GstNavigation key input events in osxvideosink compatible with x(v)imagesink ones
Wim Taymans [Tue, 29 Jan 2013 09:30:32 +0000 (10:30 +0100)]
rtpsession: avoid '...is used uninitialized'
Youness Alaoui [Wed, 9 Jan 2013 18:24:49 +0000 (13:24 -0500)]
qtdemux: set interleaved layout correctly for LPCM audio
https://bugzilla.gnome.org/show_bug.cgi?id=663458
Youness Alaoui [Wed, 9 Jan 2013 01:45:21 +0000 (20:45 -0500)]
qtdemux: add support for LPCM fourcc (uncompressed audio in Quicktime7)
https://bugzilla.gnome.org/show_bug.cgi?id=663458
Youness Alaoui [Wed, 9 Jan 2013 01:42:35 +0000 (20:42 -0500)]
qtdemux: print all debug for sound sample description v2
https://bugzilla.gnome.org/show_bug.cgi?id=663458
Youness Alaoui [Wed, 9 Jan 2013 01:14:17 +0000 (20:14 -0500)]
qtdemux: sound sample description v2 doesn't override samples_per_packet
https://bugzilla.gnome.org/show_bug.cgi?id=663458
Youness Alaoui [Wed, 9 Jan 2013 00:57:50 +0000 (19:57 -0500)]
qtdemux: pass stsd data to qtdemux_audio_caps()
We will need that later for LPCM format support. Disable
QDM2 parsing of stsd data which dead code before as well
because data was always NULL.
https://bugzilla.gnome.org/show_bug.cgi?id=663458
Youness Alaoui [Wed, 9 Jan 2013 00:56:46 +0000 (19:56 -0500)]
qtdemux: add len check for sound sample descriptions v1 and v2
https://bugzilla.gnome.org/show_bug.cgi?id=663458
Tim-Philipp Müller [Mon, 28 Jan 2013 22:42:25 +0000 (22:42 +0000)]
rtpmanager: use C89-style comments
Olivier Crête [Mon, 28 Jan 2013 23:06:15 +0000 (18:06 -0500)]
gstrtpsession: Fix double-declared variable
Olivier Crête [Mon, 28 Jan 2013 22:58:20 +0000 (17:58 -0500)]
rtp: Fix compilation errors in previous patches
Haakon Sporsheim [Thu, 28 Apr 2011 20:59:28 +0000 (22:59 +0200)]
rtpsession: Ensure MT safe event handling and plug event leak.
https://bugzilla.gnome.org/show_bug.cgi?id=667826
Idar Tollefsen [Mon, 17 Oct 2011 21:45:37 +0000 (23:45 +0200)]
rtpsession: mt-safe event-push
By taking a ref of the sink-pad under lock, it won't dissappear
while the push is taking place
https://bugzilla.gnome.org/show_bug.cgi?id=667816
Pascal Buhler [Wed, 4 Jan 2012 09:29:45 +0000 (10:29 +0100)]
rtpssrcdemux: Safely push on pads that might be removed due to a RTCP BYE
https://bugzilla.gnome.org/show_bug.cgi?id=667815
Stefan Sauer [Mon, 28 Jan 2013 19:42:26 +0000 (20:42 +0100)]
Automatic update of common submodule
From a942293 to 2de221c
Tim-Philipp Müller [Mon, 28 Jan 2013 11:54:54 +0000 (11:54 +0000)]
sbcparse: init some variables to avoid bogus compiler warnings
Wim Taymans [Mon, 28 Jan 2013 11:41:04 +0000 (12:41 +0100)]
rtpdepay: remove payload type restrictions
Remove the pt restrictions for all the depayloaders that have an
encoding-name. We can use this to autoplug decoders.
Remove the encoding-name for all the payloaders with a fixed payload
type.
We now either have an encoding-name or a pt in the sinkpad caps of
a depayloader.
See https://bugzilla.gnome.org/show_bug.cgi?id=639292
Marc Leeman [Mon, 28 Jan 2013 11:23:41 +0000 (12:23 +0100)]
rtp: remove payload requirements from selected depayloaders
encoding name is required in the caps and is a better fit for autoplugging than
the pt value. Hardware manufacturers have a bad habit of skimming through RFCs
and in this case; use unassigned numbers for encoders instead of dynamic
numbers.
In essence, this patch will add support for a lot of Bosch hardware encoders
without breaking autoplugging.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=639292
B.Prathibha [Sun, 27 Jan 2013 04:47:59 +0000 (10:17 +0530)]
tests: use g_timeout_add_seconds instead of g_timeout_add
https://bugzilla.gnome.org/show_bug.cgi?id=692615
Mark Nauwelaerts [Sun, 27 Jan 2013 11:54:15 +0000 (12:54 +0100)]
qtdemux: push mode: only parse moov 1 once
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=691570
Tim-Philipp Müller [Thu, 24 Jan 2013 21:08:51 +0000 (21:08 +0000)]
qtmux: set language to 'undefined' instead of English by default
Olivier Crête [Thu, 24 Jan 2013 02:35:25 +0000 (21:35 -0500)]
ximagesrc: Set the pixel aspect ratio correctly in the caps
Sjoerd Simons [Tue, 8 Jan 2013 07:56:45 +0000 (08:56 +0100)]
v4l2: Re-enable prepare-format emission
With the port to gstreamer 1.0 the prepare-format signal stopped being
emitted. Start emitting this again for use in uvch264src. While there
change the emission to include the caps for extra flexibility instead of
fource, width, height.
https://bugzilla.gnome.org/show_bug.cgi?id=692042
Benjamin Gaignard [Tue, 22 Jan 2013 17:12:10 +0000 (18:12 +0100)]
autogen.sh: allow calling from out-of-tree
Signed-off-by: Benjamin Gaignard <benjamin.gaignard@st.com>
https://bugzilla.gnome.org/show_bug.cgi?id=692309
Mark Nauwelaerts [Tue, 22 Jan 2013 18:26:09 +0000 (19:26 +0100)]
audioparsers: sbc: fix bogus compiler warning
gst-plugins-good/gst/audioparsers/gstsbcparse.c: In function 'gst_sbc_parse_handle_frame':
gst-plugins-good/gst/audioparsers/gstsbcparse.c:210:32: error: 'ch_mode' may be used uninitialized i
Tim-Philipp Müller [Sat, 19 Jan 2013 13:27:48 +0000 (13:27 +0000)]
pulsesink: don't error out if pa_stream_proplist_update() with new tags fails
Shouldn't really happen these days, but if it does, it's not really
a problem either.
https://bugzilla.gnome.org/show_bug.cgi?id=656068
Tim-Philipp Müller [Wed, 16 Jan 2013 18:01:23 +0000 (18:01 +0000)]
tests: skip souphttpsrc tests if there is no local http server to use
Skip tests if the server couldn't be started or we can't connect
to it for some reason (e.g. draconic build bot environments).
Thijs Vermeir [Wed, 16 Jan 2013 13:32:56 +0000 (14:32 +0100)]
autoparsers: use appropriate printf format for gsize
Martin Pitt [Tue, 15 Jan 2013 14:05:43 +0000 (15:05 +0100)]
tests: use _1_0 variants for the various registry variables
These override the variants without version suffix. Makes 'make check' work
properly in environments that set the suffixed variant for 1.0, such as
jhbuild.
Alexey Chernov [Fri, 11 Jan 2013 15:24:43 +0000 (19:24 +0400)]
osxvideosink: Fix crash in osxvideosink with external window output
Alexey Chernov [Wed, 16 Jan 2013 08:04:59 +0000 (12:04 +0400)]
osxvideosink: Make GstGLView propagate input events to its parent view
Fixes bug #691832
Tim-Philipp Müller [Wed, 16 Jan 2013 10:19:36 +0000 (10:19 +0000)]
rtpsbcpay: update some fields in the caps to their new name
and to match the parser. "mode" got renamed to "channel-mode"
and "allocation" to "allocation-method".
Tim-Philipp Müller [Tue, 15 Jan 2013 17:44:33 +0000 (17:44 +0000)]
docs: add sbcparse and rtpsbcpay to plugin docs
Tim-Philipp Müller [Tue, 15 Jan 2013 17:38:24 +0000 (17:38 +0000)]
audioparsers: add SBC audio parser
From-scratch rewrite, the bluez one was useless and broken.
https://bugzilla.gnome.org/show_bug.cgi?id=690582
Tim-Philipp Müller [Tue, 15 Jan 2013 15:05:04 +0000 (15:05 +0000)]
Automatic update of common submodule
From a72faea to a942293
Tim-Philipp Müller [Thu, 10 Jan 2013 12:38:13 +0000 (12:38 +0000)]
rtp: import rtpsbcpay from bluez and port to 1.0
Compiles, but not tested yet (sbc elements still need to be ported).
https://bugzilla.gnome.org/show_bug.cgi?id=690582
Marcel Holtmann [Mon, 14 Feb 2011 01:51:45 +0000 (17:51 -0800)]
rtpsbcpay: Remove workaround for compiler warnings
Marcel Holtmann [Wed, 19 May 2010 14:59:30 +0000 (16:59 +0200)]
rtpsbcpay: Add pragma based workaround for GStreamer warnings
Marcel Holtmann [Sat, 2 Jan 2010 01:08:17 +0000 (17:08 -0800)]
rtpsbcpay: Update copyright information
Marcel Holtmann [Thu, 29 Jan 2009 23:31:15 +0000 (00:31 +0100)]
rtpsbcpay: Fix signed/unsigned comparison issue within GStreamer plugin
Marcel Holtmann [Thu, 1 Jan 2009 18:33:20 +0000 (19:33 +0100)]
rtpsbcpay: Update copyright information
Marcel Holtmann [Tue, 23 Dec 2008 04:25:50 +0000 (05:25 +0100)]
rtpsbcpay: First attempt in fixing compiler warnings (still needs cleanup)
Johan Hedberg [Sat, 20 Dec 2008 19:42:49 +0000 (21:42 +0200)]
rtpsbcpay: More coding style fixes
Luiz Augusto von Dentz [Fri, 29 Feb 2008 19:37:15 +0000 (19:37 +0000)]
rtpsbcpay: Remove possible extra memcpy for gstreamer plugin.
Luiz Augusto von Dentz [Thu, 28 Feb 2008 19:38:53 +0000 (19:38 +0000)]
rtpsbcpay: Fix bug sending empty packages and remove a buffer copy.
Luiz Augusto von Dentz [Wed, 20 Feb 2008 13:37:00 +0000 (13:37 +0000)]
rtpsbcpay: Fix runtime warnings of gstreamer plugin.
Luiz Augusto von Dentz [Tue, 19 Feb 2008 19:49:24 +0000 (19:49 +0000)]
rtpsbcpay: Update gstreamer plugin to use new sbc API.
Marcel Holtmann [Sat, 2 Feb 2008 03:37:05 +0000 (03:37 +0000)]
rtpsbcpay: Update copyright information
Luiz Augusto von Dentz [Wed, 30 Jan 2008 14:21:43 +0000 (14:21 +0000)]
rtpsbcpay: Fixes gstreamer caps and code cleanup.
Luiz Augusto von Dentz [Thu, 24 Jan 2008 14:25:29 +0000 (14:25 +0000)]
rtpsbcpay: Fix gtreamer payloader sending fragmented frames.
Luiz Augusto von Dentz [Wed, 23 Jan 2008 19:17:33 +0000 (19:17 +0000)]
rtpsbcpay: Fix use of gstreamer plugin with rhythmbox and banshee and rtp timestamps.
Luiz Augusto von Dentz [Wed, 23 Jan 2008 13:14:02 +0000 (13:14 +0000)]
rtpsbcpay: Make a2dpsink to act like a bin and split the payloader.
Wim Taymans [Tue, 8 Jan 2013 15:27:42 +0000 (16:27 +0100)]
rtp: small improvements
Wim Taymans [Mon, 7 Jan 2013 14:50:33 +0000 (15:50 +0100)]
jitterbuffer: refactor handle sync code
Move the code that combines the last SR packet and the current jitterbuffer sync
values into a sync structure, into its own function. We want to reuse this bit
later.
Wim Taymans [Mon, 7 Jan 2013 14:45:10 +0000 (15:45 +0100)]
rtp: include downstream latency in SR calculations
When we make a mapping between an RTP timestamp and an NTP timestamp, include
the downstream latency applied to the sinks. This makes it possible to have
both sinks run with different latencies and still have correct sync on the
client. It also is more correct because the RTP timestamp in the SR report will
actually correspond more closely to the NTP time it was sent on the server.
For pipelines with high latency on the sender side, this actually allows a
GStreamer receiver to perform synchronisation instead of dropping the RTCP
packets.
Wim Taymans [Mon, 7 Jan 2013 13:25:14 +0000 (14:25 +0100)]
rtpsession: don't cast event functions
There is no need to cast the event functions and only causes problems later when
we change the signature later and things silently compiles wrong code.
Wim Taymans [Mon, 7 Jan 2013 13:23:34 +0000 (14:23 +0100)]
rtp: more debug
Wim Taymans [Mon, 7 Jan 2013 13:22:48 +0000 (14:22 +0100)]
rtpsession: improve debug
Tim-Philipp Müller [Wed, 2 Jan 2013 00:03:27 +0000 (00:03 +0000)]
udpsrc: sanity check size of available packet data for reading to avoid memory waste
On Windows and OS/X, _get_available_bytes() may not return the size
of the next pending packet, but the size of all pending packets in
the kernel-side buffer, which might be rather large depending on
configuration. Sanity-check the size returned by _get_available_bytes()
to make sure we never allocate more memory than the max. size for
a packet, if it's an IPv4 socket.
https://bugzilla.gnome.org/show_bug.cgi?id=610364
Robert Krakora [Fri, 4 Jan 2013 09:03:32 +0000 (10:03 +0100)]
v4l2: Also handle the new ENOENT return value of VIDIOC_QUERYCTRL
https://bugzilla.gnome.org/show_bug.cgi?id=691098
Tim-Philipp Müller [Tue, 1 Jan 2013 19:14:36 +0000 (19:14 +0000)]
tests: add test for souphttpsrc error handling with data
https://bugzilla.gnome.org/show_bug.cgi?id=678429
Norbert Waschbuesch [Fri, 22 Jun 2012 21:56:52 +0000 (21:56 +0000)]
souphttpsrc: error out properly when receiving data along with an error status
When receiving an error code from the http server, such as 404,
data might be sent along with it, like a web page. We don't want
to output that data in this case, and we also want to pass the
FLOW_ERROR return back to the base class, so it can stop properly.
https://bugzilla.gnome.org/show_bug.cgi?id=678429