This is necessary because the sinks don't notice the group switches
and the decoders/demuxers have a different running time than the
sinks.
Fixes bug #537050.
In some cases (all buffers dropped by a parser) a decodebin2
chain might receive an EOS before it gets enough data to
expose a decoded pad. In the case that no streams can expose
a pad we should error out instead of hang.
Fixes#542758
Just counting how many messages were sent and how many were received
is not good enough because they might've been duplicated (e.g. by the
visualization audio tee). Comparing the sequence numbers should give
better results in that case.
Otherwise the async state change from READY->PAUSED of the
uridecodebins will take playbin2 from PLAYING->PAUSED again
during gapless group switches.
Fixes bug #602000.
When a decodebin2 receives no-more-pads of a group it
can set that group's multiqueue buffering thresholds to
'playing' buffering method, avoiding that it buffers
too long and cause problems when using with queue2.
See the associated bug for details.
Fixes#600787
During a group switch return the cached duration of the old group
because the old group still didn't finish playback. If we have no
cached duration return FALSE.
Fixes bug #585969.
Make sure, to only "simulate" subtitle no-more-pads if it was still
pending and also handle errors in the subtitle pipeline as warnings
after the subtitles prerolled.
Don't set the suburidecodebin to READY after errors, handle_message
will usually be called from the streaming thread and doing that
from there is obviously not a good idea.
Now the caps property isn't set anymore for the subtitle caps
but instead in the autoplug-continue signal it is detected
if the caps belong to a supported subtitle stream.
This makes automatic use of newly installed plugins.
First of all, make sure that suburidecodebin never
errors out because of not-linked in case external subtitles
are used but then subtitles are disabled.
And then make sure that external subtitles always start from
the correct position and are not racing until EOS if they
get unselected and selected again.
This will make sure that no subparse is ever plugged and subtitleoverlay,
that subpicture streams are handled the same was as subtitles and that
subtitle renderers are used if available.
Fixes bugs #595123, #570753, #591662, #591706.
Using the object lock here can and will lead to deadlocks because
of deep-notifies of property changes: the deep-notify handler will
get the parent of objects, which will take the object lock again.
Fixes bug #600479.
Use the faster gst_element_link_pads because we know for sure the sinkpad name
and we don't need to have the function search for a suitable pad anymore.
We want to return NOT_LINKED for unselected pads but only for pads
from the normal uridecodebin. This makes sure that subtitle streams
are not raced past audio/video from decodebin2's multiqueue.
For pads from suburidecodebin OK should always be returned, otherwise
it will most likely stop with an error.
There's not much point in using GST_DEBUG_FUNCPTR with GObject
virtual functions such as get_property, set_propery, finalize and
dispose, since they'll never be used by anyone anyway. Saves a
few bytes and possibly a sixteenth of a polar bear.
Set the output caps on the srcpad before pushing the buffer because else core
will do a rather expensive check to see if we can actually accept those caps on
the srcpad.
Install a custom acceptcaps function instead of using the default expensive
check. We accept whatever downstream accepts so we pass along the acceptcaps
call to the downstream peer.
* memcmp is expensive and was being abused, reduce calling it by checking
the first byte.
* iterating one byte at at time over 64 kbites introduces a certain overhead,
therefore we now do it in chunks of 1024 bytes
And I do mean over 300 times. The average instruction call per mxf_type_find
was previously 785685 and it's now down to 2458 :)
I also renamed glib_enum_prefix to glib_gen_prefix as we also use that for the
marshallers. Also rename the rtsp-marshal.list to work with the unified prefix.
instead of printing an error that no corresponding group could
be found. no-more-pads from non-demuxer elements doesn't give
any additional information because there can only be a single srcpad.
Fixes bug #598288.
This allows partial group changes, i.e. demuxer2 in the example below
goes EOS but has a next group and audio2 stays the same.
/-- >demuxer2---->video
demuxer--- \--->audio1
\--->audio2
This now keeps track of everything that is going on, creates
a tree of chains and groups to allow "demuxer after demuxer" scenarios
and allows chained Oggs with multiple streams (needs oggdemux or playbin2 fixes).
Also document everything in detail and give a general overview of what
decodebin2 is doing at the top of the sources.
Fixes bug #596183, #563828 and #591677.
Pad blocks should never be done on external pads as outside elements
might want to use their own pad blocks on them and this will lead to
conflicts and deadlocks.
Adds a pattern with out-of-gamut colors in a checkerboard
pattern with in-gamut neighbors. Useful for checking YCbCr->RGB
color matrixing. Correct matrixing and clamping will cause the
checkerboard pattern to be invisible.
This allows using playsink from outside the playback plugin.
Add code to be able to request the sink pads using standard GStreamer API.
TODO : expose GObject properties/signals.
Add a property that makes videorate skip to the first buffer it
receives instead of padding the stream from segment start to the
first real buffer.
Fixes bug #567928.
videotestsrc rounds chroma down. This causes it to omit the last chroma
value completely for odd widths when the chroma is downsampled.
This patch special cases the last pixel to not be rounded down.
Disable headerless flac typefinder as long as it happily typefinds anything
including /dev/urandom as flac and as long as it's not particularly useful
given that such streams don't really exist in the wild.
Also fix up some comments so that gtk-doc doesn't complain about them.
This is a standard Midi file format that should be supported by
all Midi decoders and also has the mimetype audio/mid according to
the Midi specification homepage.
Fixes bug #594094.
Its purposes is mainly to avoid false positives (e.g. mp3 typefinder
reporting a 20% probability and somesuch). Won't be registered if
the gio plugin has been disabled via ./configure --disable-gio.
Also use the capsfilter if there is no src-peer as the caps constrain what
we can do. Don't create any_caps as a default, as we check for NULL to skip the
filtering. This is a (small) performance regression as we always intersect
otherwise.
g_value_set_object() increases the refcount of the sink, which is not needed
because the object should already be refcounted. Make sure this is always the
case and use g_value_take_object().
Fixes: #592884
Before, SEEK events would be sent to the video sink, which wouldn't
be linked in any way to the subtitle part of the pipeline and
subparse would never see the SEEK event. This would then seek
the audio/video but the subtitles would continue from the old
position instead.
Fixes bug #591664.
The problem with an error message is, that it will stop playback completely
while it could be that only a audio decoder plugin is missing and the video
could be played with the available plugins.
See bug #591677.
Before we had STREAM/WRONG_TYPE but it's really CORE/MISSING_PLUGIN
because a plugin is missing and nothing else is wrong.
Also make it an error instead of a warning.
Really fixes bug #591677.
Don't do fallbacks if application specified a sink element. When doing the
fallback use configured default elements instead of hardcoded linux only
elements. Improve error messages accordingly.
If a downstream element returns an error while upstream has already
put all data into queue2 (including EOS), upstream will no longer
chain into queue2, so it is up to queue2 to perform some
EOS handling / message posting in such cases. See #589991.
This later allows to handle interlaced AVPicture different than
progressive ones which is needed for horizontally subsampled YUV
formats, see bug #589242.
There are flac-in-ogg files without the usual flac packet framing
and these files just have a 4-byte fLaC ID packet as first packet.
We need to recognise the type just from these four bytes if we
want oggdemux to recognise these streams correctly.
Rename the GType of the pads of playbin's internal stream selector
element so they don't use the same type name as input-selector's
pads. Fixes#589622.
We can't call gst_element_send_event() from a streaming thread as it gets the
state lock. Instead call the send_event method directly until we have a nice API
for this in basesrc.
Fixes#588746
Keep track of the max requested position and compare this to the write position
in the temp file to get the current amount of buffered data.
Fix memleak of all incomming buffers.
Fixes#588551
We shouldn't really depend on elements from -bad for stream
selection in playbin2, so use a private copy of input-selector
until the selector plugin is ready to be moved to -base or -good.
Fixes#586356.
Differentiate subtitle streams and lyrics/cracktastic/complex streams via
the category string in the headers. This seems like a useful distinction
to make, and also seems more future-proof. See #525743.
Don't flush the file by closing and opening it but instead use g_freopen. This
avoids a deadlock in shutdown because we emit the temp-location property change
with the wrong lock held.
Fix the construction of the temporary filename construction as the application
name can be NULL and we don't want a separator between the prgname and the
template.
Add a download property that will attempt to configure queue2 into progressive
download buffering.
Make sure we only enable download buffering for quicktime and flv formats.
Add a new temp-template property so that queue2 can securely allocate a
temporary filename. Deprecate the temp-location property for setting the
location but still use it to notify the allocated temp file.
Adder can only handle one common format accross the pads. Thus one needed to add
a capsfilter afterwards and manage the caps. Now one can simply set the caps on
the property.
If READY->PAUSED failed in the source element we would've swapped
the current and next group already. To allow READY->PAUSED to succeed
after the first failure we have to swap the current and next group
back again. This also ensure that we're again in the same state
as before the failed state change and not at the next group.
This was especially a problem for playbin2 pipelines that use the
new mounting support in giosrc as the source would fail for READY->PAUSED
the first time, the application mounts the location and then tries
to go READY->PAUSED again (and this time it would succeed).
Fixes bug #588078.
This ensures that collectpads' cookie is properly updated so that when the streaming
threads will restart and be checking for the flushing status of all pads there will
be no inconsistent state.
This patch adds support for stationary white Gaussian noise.
The Box-Muller algorithm is used to generate pairs of independent
normally-distributed random numbers.
Fixes bug #586519.
When a seek failed upstream, make sure the adder sinkpad is set unflushing again
so that streaming can continue.
We only have a pending segment when we flushed.
Set the flush_stop_pending flag inside the appropriate locks and before we
attempt to perform the upstream seek.
Add some more comments.
Use the right lock to protect the flags in flush_stop.
See #585708
Set the target state of the newly added uridecodebins to somthing else that
PAUSED so that we keep their state in sync with the playsink state.
Fixes#585268
At least do the fix to sent the flush_stop when seeking failed to ensure we
keep no pads flushing. before it was send when the seeking worked which is just
plain wrong and was not the intention.
When no flush-stop has been sent by upstream, we have to send one ourselves to
continue playback. Do this as soon as the collect function is called instead of
after we possibly pushed segment events (that got then flushed out)
uridecodebin expects the passed connection-speed value in kbps, so we
need to divide the value stored in bps by 1000. Also, lower the upper
limit on the properties to the value that we can actually store in our
internal guint (which is plenty high enough)
Be even less restrictive in what we accept for .srt timestamps when
typefinding and parsing subrip subtitles and add a unit test for
the 'new' format. Fixes#585197.
When we are probing for streams, we want to set the queue size in such a way
that we can scan a maximum amount of data without consuming too much memory.
Therefore, remove the time limit on the queue and only stop scanning after 2MB
of data.
See #584104.
Recognise PGS subpicture streams and connect them to the SPU pad
in playsink. Unfortunately this fails badly with negotiation errors
if the SPU is not recent enough to support the stream. I'm not sure
how to add format negotiation in yet.
When using an audio sink without a "volume" property, volume control
would only work for the first song. For the next song, we'd try to
re-use the existing audio chain, but inadvertently set chain->volume
to NULL instead of to the existing volume element.
playbin2 inadvertently used autoaudiosink and autovideosink up to now,
since it would overwrite the sinks configured via the "audio-sink"
and "video-sink" properties with the stream-specific group sinks when
configuring the outputs. Those are usually NULL however, so that would
overwrite the configured sinks with NULL which makes playbin2 then
default to the auto sinks. Fix this by keeping a reference to each
configured sink in playbin2 and setting up the right sinks depending
on whether there is a stream-specific sink or not.
Fixes#584020.
Use two flags to remember volume/mute changes at times when we don't have the
audiochain yet (e.g. construction). Only set values when they were actualy
changed. This makes pulseaudio's stream restore functional.
Adder was relying that something else sends a flush stop. When using adder with
a livesource it was not getting a flush_stop and thus all pads downstream where
keept flushing. Mark a pending flush_stop and send it when we are working on
the new segment back in the streaming thread.
Add a queue2 after the raw output pads of certain sources such as those for uris
like cdda://
No tuning of the queue is done yet as the defaults seem to work fine for me.
Fixes#582528
The enum nick should be 'sine-table', not 'sine table'. Technically this is
an API/ABI change I guess, but anyone who was using this and didn't report
it deserves this.
Handle buffers with -1 timestamps better by keeping track of the en time of the
previous buffer and assuming the -1 timestamp buffer goes right after the
previous one.
when we have two buffers that are equally good, output the oldest buffer once to
minimize latency.
don't try to calculate latency when the input framerate is unknown.
Keep track of the autoplugged custom sinks and configure them in the playsink
element when we have collected all streams.
Also make sure that we only select one custom sink.
When unreffing the internal sink, we don't need to change the state to NULL.
mp3_type_find could suggest already when only a single valid header
was found, if it ran out of data before the end of the next frame.
Therefore, ignore the last found frame if it was incomplete.
Fixes bug #579692.
Make playsink go async to the PAUSED state instead of relying on uridecodebin
for async behaviour in playbin. This solves some problems (mainly with DVD)
where the pipeline would go to PLAYING before preroll completed, failing to
select the audiosink clock.
Fixes#581727
When calculating the input/output buffer sizes in the transform_size function,
take the number of channels into account, so we don't end up calculating
a buffer size that only contains a partial number of audio frames.
Also, when going from output size to input size, round down rather than
up, so as to calculate the minimum number of samples that *might* yield
a buffer of the intended destination size.
Fixes: #580470 and #580952
When videorate duplicates a buffer with a DISCONT flag, it copies the discont on
the first pushed buffer but fails to clear it for subsequent buffers. This
causes theoraenc!oggmux and possibly other elements to consider this a discont
stream.
Fix videorate to produce discont as the first buffer and after a flushing seek.
Fixes#580271.
The 2s limit is way too small for a lot of files (which have an interleave
in time of between 3 and 5s). Instead, leave it to the initial 5s value
and reduce the other limits (allowing us to stay memory-efficient).
First check the pad caps if they are raw before setting the raw_decoding_mode to
TRUE. Fixes playback of transport streams and other streams that require large
queues.
Fixes#579734
Adds a new property in multifdsink, resend-streamheader.
If this property is false, the multifdsink will not send the streamheader if
there's already one set for a particular client.
There are some formats in which every stream needs to start with a certain
blob, but you can't inject this blob at leisure. If the producer wants to
change the blob in question and sets in as the streamheader on the outgoing
buffers' caps, new clients of multifdsink will get the new streamheader, but
old clients will break, because they'll see the blob in the middle of the
stream.
The property is true by default, so existing code will not see any difference.
Fixes#578118.
Add a property to disable listening to client writes. This property is usefull
when other code will deal with reading from the client socket.
API: GstMultiFdSink::handle-read property
Clear the target of our ghostpads before we remove the pad from the element.
This to make sure that the internal pad is not left linked to whatever pad we
were ghosted to. This should only be a problem when we leak the ghostpads.
Also release our subpicture pads.
Fixes#577288.
Raw decoding mode removes almost all buffering in video and audio queues
when a source providing already decoded video/audio is detected, on the
possibly bogus assumption that such a source should provide sufficient
internal queueing. Fixes playback on some DVDs, and improves it
on all.
If one side has a preference for a particular sample rate or set of sample rates, we
should honour this in the caps we advertise and transform to and from, so that elements
actually know about the other side's sample rate preference and can negotiate to it
if supported. Also add unit test for this.
This prevents valgrind warnings when accessing the "x" parts
of xRGB and friends in other elements that handle (and can handle)
xRGB like ARGB (for example videoscale).
When reusing playbin with visualisations, reset the async property on the video
sink because some sinks might dynamically recreate their sinks.
Fixes#576188
When we have the textpad configured, enable and disable the subtitles by setting
the silent flag on the overlay element instead of trying to remove elements.
See #576187
Updated the examples in the README to actually work. Add them to api docs. Tests
the api-docs and fix the section names to make the docs actualy show up.
The example for "tcpserversrc" needs review (might be an element bug).
Link after doing the state change and unlink before shutting down. Makes the
window for causing races in toggling the visualisations smaller.
See #576187.
Remove the group GCond that we used for waiting for groups to finish because we
use pad blocking on the selectors and counters instead for waiting for the
groups to complete.
remove the obsolete about_to_finish variable set while emiting the
about-to-finish signal and fix some old comments.
We don't need to take the playbin lock when querying the uridecodebin.
When we make a group connected to a demuxer, keep an extra dynamic refcount for
the group which is only decremented when no_more_pads or a multiqueue overrun is
detected. This way we avoid a race between exposing the group while more dynamic
refs are added from new pads.
Fixes#575588.
Sync the state of the newly added chains to the state of the parent sink element
to avoid lost async-start messages. Fixes cdda:// async-done message storm.
When streams are not selected in the selector, return NOT_LINKED so that
upstream elements can skip decoding. Only do this for audio and video pads
because for text streams the overhead is smaller and they could come from
external files.
Set the custom sink async=FALSE to not make it participate in preroll because we
are dealing with sparse streams.
Try to set sync=TRUE on the custom text sink.
Release the shutdown lock when we wait for other groups to complete or else we
have a deadlock when the other group completes and tries to grab the shutdown
lock.
Fixes#575550.
The flac frame header typefinder overstates the likelihood of a match, leading
to false positives with e.g. aac streams and PDF files. Reduce probabilty
returned from LIKELY to POSSIBLE for the frame header matchin code.
Fixes#574939.
Detect more variations and also bail out in more cases where the values
don't make sense. Furthermore, add width/height and bpp to the caps,
because we can.
Add property to playbin2 to configure a custom sink that receives the raw
subtitle buffers instead of using a textoverlay.
Improve the property finding code to make it more usable.
Use property find code to find async properties in custom sinks that are bins.
Improve text overlay code to gracefully handle missing elements.
Use scan context for initial peek as well. Peek 6 bytes in the initial
peek rather than 5 bytes, to match the length of the memcmp we're doing
on that data later. Return immediately when we found caps from looking
at the beginning of the data - no point in continuing to scan the next
64kB for something matching a frame header.
Disconnect the notify::caps signal in our callback (it'll be re-added
if we're not, in fact, finished getting complete caps). Ensures that
caps changes mid-stream (e.g. from an mp3 that changes from
stereo->mono mid-file) don't cause us to try to add a new pad.
Make it possible to request a flushing pad from the playsink. We can eventually
use these flushing pads to quickly terminate the dataflow when we are shutting
down.
Release the group lock while we perform the state changes on the uridecodebins
because that might trigger callbacks that we need to handle with the group lock
taken. Avoids a possible deadly embrace in some id3/flac files.
Fixes#567396.
When setting the quality/filter-length while PLAYING the
resampling context will be destroyed and created again in
some cases, which will cause crashes in the transform function
if it's called at that time.
Rather than only checking for volume property on the audio sink
directly, recursively look for it on sinks within it (if it's a bin).
Allows use of sink-as-volume-control where the application has supplied
an audio-sink bin that includes a real audio sink internally.
Don't keep extra references to volume and mute elements; we don't need
to do so.
Ensure we unref pads that we have references to, and release request
pads.
Because core now supports typefindfactories without a typefind function we can
register a factory fo GSM that will --if all else fails-- assume the file is a
GSM file based on the registered extension.
Fixes#566661.
We can use gst_element_link_pads() instead of the more generic
gst_element_link() function because we know the pads. This saves some cycles
because the more generic function needs to search for possible compatible caps
etc.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
(gst_play_bin_set_uri), (gst_play_bin_set_suburi),
(no_more_pads_cb), (drained_cb), (group_set_locked_state_unlocked),
(activate_group), (deactivate_group), (groups_set_locked_state),
(gst_play_bin_change_state):
Fix some comments and docs.
Post an error message when we fail to link the selector to the sink.
Remove pushing of EOS, this seems unneeded.
Lock the state of deactivated groups so that they don't accidentally
reactivate when the playbin2 state changes.
Reuse uridecodebins.
Unlock and relock state of groups when playbin goes to NULL.
Fixes#566654.
Fixes#566341.
* gst/playback/gsturidecodebin.c: (pad_removed_cb), (type_found):
Only do something in the pad removed callback when we are dealing with
our sourcepads because the sinkpads don't have a ghostpad.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (notify_source_cb), (activate_group):
Disconnect signal handlers before destroying a previous decodebin so
that we don't end up causing deadlocks. Fixes#566586.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init), (gst_audio_test_src_init),
(gst_audio_test_src_check_get_range),
(gst_audio_test_src_set_property),
(gst_audio_test_src_get_property):
* gst/audiotestsrc/gstaudiotestsrc.h:
Add property to control pull/push based scheduling.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (no_more_pads_cb):
Add some debug info.
* gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
(gst_play_sink_reconfigure), (gst_play_sink_request_pad),
(gst_play_sink_release_pad):
Add some more debug info.
Reconfigure the audio chain when we switch between raw and encoded audio
in gapless playback.
Original commit message from CVS:
* gst/adder/Makefile.am:
* gst/adder/gstadder.c:
Cleanup variable names to make the adder-loop easier to understand.
Also try to use liboil to spee it up, but ifdef it out as it does not
make any change for me (Intel pentim M (sse,sse2) please try on other
systems).
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* gst/tcp/gsttcpclientsink.c:
* gst/tcp/gsttcpclientsrc.c:
* gst/tcp/gsttcpserversrc.c:
Add minimal docs to make the remaining tcp elements show up.
Fixes#564139.
Original commit message from CVS:
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_finalize):
Free the factory array when finalizing.
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_base_init):
Use a GstStaticPadTemplate since the src pad caps are fixed.
Original commit message from CVS:
* gst/subparse/samiparse.c: (sami_context_push_state),
(sami_context_pop_state), (start_sami_element), (end_sami_element):
Some versions of libxml seem to be very picky as to strict formatting
of the input and never 'close' the final </body> tag.
In order to fix that bad behaviour, we trigger the flushing of
remaining data on both </body> and </sami>.
Fixes#557365
Original commit message from CVS:
Patch by: Guillaume Emont <guillaume at fluendo dot com>
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Add typefinders for MS Word files and OS X .DS_Store files to
prevent them to be recognized as MPEG files. Fixes bug #564098.
Original commit message from CVS:
* gst/playback/gstplaysink.c: (gen_audio_chain),
(gst_play_sink_reconfigure):
Add some more debug info.
Fix linking of just an encoded sink.
Handle failure to create a sink chain more gracefully than crashing.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (pad_added_cb):
Error out with a missing-plugin error when the input-selector was not
found.
* gst/playback/gstplaysink.c: (gst_play_sink_reconfigure):
Indentation.
Original commit message from CVS:
* gst/playback/gstplaysink.c: (gst_play_sink_class_init),
(gst_play_sink_dispose), (gst_play_sink_finalize), (try_element),
(gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
(gst_play_sink_send_event), (gst_play_sink_change_state):
Use G_DEFINE_TYPE.
Try to set the selected sink to READY before using it. This will allow
for detection of incompatible formats sooner.
Don't cause a fatal error when conversion elements are missing but post
a missing-element message and a warning instead because things might
still link and run fine.
Simplyfy the construction of audio and video sink chains.
Original commit message from CVS:
Patch by: Luis Menina <liberforce at freeside dot fr>
* gst-libs/gst/floatcast/floatcast.h:
* gst/typefind/gsttypefindfunctions.c:
Include glib.h instead of a specific GLib header. Including single
GLib headers is deprecated. Fixes bug #563904.
Original commit message from CVS:
* gst/volume/gstvolume.c: (gst_volume_class_init),
(volume_before_transform), (volume_transform_ip):
Use new basetransform vmethod to reconfigure the dynamic properties and
any pending volume/mute changes. Fixes#563508.
Original commit message from CVS:
* gst/playback/gstdecodebin.c:
* gst/playback/gstdecodebin2.c:
Add basic docs to decodebin and link to decodebin from decodebin2.
Original commit message from CVS:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/inspect/plugin-videorate.xml:
* gst/speexresample/gstspeexresample.c:
Update documentation of speexresample for the new element name.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c: (plugin_init):
Update the debug category from speex_resample to audioresample.
Original commit message from CVS:
* gst/playback/gstplaybin2.c:
Add notification of current stream. Add ability to configure buffer
sizes.
* gst/playback/gsturidecodebin.c:
Add ability to configure buffer sizes for streaming mode.
Bug #561734.
Original commit message from CVS:
* gst/playback/gstplaysink.c: (gen_audio_chain):
Don't post an error when we can't configure the volume but post a
warning instead. Fixes#561780.
Original commit message from CVS:
Patch by: Jonathan Rosser <jonathan.rosser@rd.bbc.co.uk>
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
Add a zone plate pattern generator based on BBC R&D Report
1978/23 (yeah *that* 1978). Try 'videotestsrc pattern=zone-plate
kx2=20 ky2=20 kt=1'.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_class_init), (gst_speex_resample_set_property),
(gst_speex_resample_get_property):
Add a "filter-length" property that maps to the quality values
for compatibilty with audioresample.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
If the top-level type of the stream is plain text, don't try to decode
it, matching behaviour of decodebin.
* gst/playback/gstplaysink.c:
If we fail to generate a text chain (e.g. due to missing optional
plugins), don't crash.
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
Add "colorspec" property, specifying whether to generate BT.601
or BT.709 video. This only affects YCbCr values, not RGB, since
if you're generating a 709 test pattern, presumably you want
709 RGB primaries, not 601. Also add "smpte75" pattern, which
uses 75% colors instead of 100%, since this is often more useful
for testing (and also follows the SMPTE EG-1 guideline).
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
Guard against a NULL dereference I somehow encountered -
with a FLUSH_STOP arriving either before basetransform _start(),
or after _stop().
* gst/typefind/gsttypefindfunctions.c:
Make sure we never jump backwards when typefinding corrupt mov files.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (jp2_type_find),
(plugin_init):
Improve typefinding of ISO JPEG2000 mime types.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (deactivate_group):
don't try to unlink the selector sinkpad when we don't have it yet. This
can happen if an error occured before the group was complete.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (activate_group):
Catch state change errors and stop from the uridecodebin elements
instead of trying to continue in vain.
Original commit message from CVS:
* gst/playback/gstplaysink.c: (gst_play_sink_reconfigure):
Don't try to do crazy things when we only have a text pad without a
video pad. Fixes#559478.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_update_real_volume),
(gst_volume_set_volume), (gst_volume_get_volume),
(gst_volume_set_mute), (gst_volume_init), (volume_setup),
(volume_transform_ip), (volume_update_mute),
(volume_update_volume), (volume_get_property):
* gst/volume/gstvolume.h:
Keep negotiated state in a separate variable.
Protect the volume and mute properties with the object lock.
Protect modifying the transform with the transform lock.
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps):
Only convert caps to string when debug is enabled.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_convert_buffer):
Add TODO at the top of the file for enabling SSE/ARM specific
optimizations and choosing the fastest implementation at runtime.
Add g_assert_not_reached() at two places that should really never
be reached.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_check_discont):
Fix format string and arguments.
* gst/speexresample/resample_sse.h:
Add missing file.
Original commit message from CVS:
* gst/speexresample/Makefile.am:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_base_init), (gst_speex_resample_get_funcs),
(gst_speex_resample_convert_buffer), (_benchmark_int_float),
(_benchmark_int_int), (_benchmark_integer_resampling),
(plugin_init):
* gst/speexresample/gstspeexresample.h:
* gst/speexresample/resample.c:
* gst/speexresample/speex_resampler_double.c:
* gst/speexresample/speex_resampler_float.c:
* gst/speexresample/speex_resampler_int.c:
* gst/speexresample/speex_resampler_wrapper.h:
Add missing headers to Makefile.am.
Update copyright, years and my mail address.
Benchmark the integer resampling implementation against the
float implementation and use the faster one for 8/16 bit integer
input. On most recent systems the floating point version is faster.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_convert_buffer):
The length for the buffer conversion function is the number of
audio frames, i.e. we need to multiply it by the number of channels
to get the number of values. Also spotted by the unit test after
running in valgrind.
Original commit message from CVS:
* tests/check/elements/speexresample.c: (element_message_cb),
(eos_message_cb), (test_pipeline), (GST_START_TEST),
(speexresample_suite):
Add pipeline unit tests for testing all supported formats with
up/downsampling and different in/outrates.
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_push_drain), (gst_speex_resample_process):
* gst/speexresample/speex_resampler_wrapper.h:
Fix bugs identified by the testsuite.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_stop),
(gst_speex_resample_get_funcs),
(gst_speex_resample_transform_size),
(gst_speex_resample_convert_buffer),
(gst_speex_resample_push_drain), (gst_speex_resample_process):
* gst/speexresample/gstspeexresample.h:
* gst/speexresample/speex_resampler_wrapper.h:
Add support for int8, int24 and int32 input by converting internally
to/from int16 or double.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_start),
(gst_speex_resample_get_unit_size),
(gst_speex_resample_push_drain), (gst_speex_resample_event),
(gst_speex_resample_check_discont), (gst_speex_resample_process),
(gst_speex_resample_transform):
* gst/speexresample/gstspeexresample.h:
Rewrite timestamp tracking to make it more robust and guarantee
a continous stream.
* tests/check/Makefile.am:
* tests/check/elements/speexresample.c: (setup_speexresample),
(cleanup_speexresample), (fail_unless_perfect_stream),
(test_perfect_stream_instance), (GST_START_TEST),
(test_discont_stream_instance), (live_switch_alloc_only_48000),
(live_switch_get_sink_caps), (live_switch_push),
(speexresample_suite):
Add unit tests for speexresample based on the audioresample unit tests.
Original commit message from CVS:
* gst/speexresample/gstspeexresample.c:
(gst_speex_resample_class_init), (gst_speex_resample_fixate_caps),
(gst_speex_resample_process):
Fixate to the nearest supported rate instead of the first one.
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
(gst_audioresample_class_init), (audioresample_fixate_caps):
Fixate the rate to the nearest supported rate instead of
the first one. Fixes bug #549510.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mxf_type_find):
Improve MXF typefinding a bit by searching for a header partition
pack instead of just a general partition pack and checking more
bytes for valid values.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init):
Set the default blocksize to -1 because we will then use the configured
samplesperbuffer to create our output buffer.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (flac_type_find):
Don't forget to advance the offset of what we're matching against, else
we end up in a forever loop.
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (gst_subparse_type_find):
Improve typefinding a bit. If we don't have a Unicode charset
try GST_SUBTITLE_ENCODING and otherwise try ISO-8859-15.
Original commit message from CVS:
* gst/subparse/Makefile.am:
* gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
(gst_sub_parse_class_init), (gst_sub_parse_init),
(gst_convert_to_utf8), (detect_encoding), (convert_encoding),
(get_next_line), (gst_sub_parse_data_format_autodetect),
(feed_textbuf), (handle_buffer), (gst_sub_parse_change_state),
(gst_subparse_type_find):
* gst/subparse/gstsubparse.h:
Add support for UTF16/UTF32 subtitles as long as the first bytes of
the first buffer contain the BOM. This also adds support for other
encodings that allow NUL bytes via the encoding property.
Fixes bugs #552237 and #456788.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (flac_type_find):
For looking at the 4th byte we have to get 4 bytes of course
and not 3.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (flac_type_find):
Improve FLAC-without-headers typefinding by looking at most of the
frame header and checking if invalid values are used. Should prevent
quite some false positives compared to the old version which only
check if the first 14 bits are set.
Original commit message from CVS:
* gst/subparse/gstsubparse.c:
(gst_sub_parse_data_format_autodetect), (handle_buffer),
(gst_sub_parse_change_state):
* gst/subparse/gstsubparse.h:
* tests/check/elements/subparse.c: (GST_START_TEST):
Add support for subtitle files with UTF-8 BOM at the beginning
by simple stripping it from the first line before passing it
to any parsing code. Fixes bug #555257 and playback of files
created by Gnome Subtitles.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
(gst_audio_test_src_class_init), (gst_audio_test_src_init),
(gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
(gst_audio_test_src_start), (gst_audio_test_src_stop),
(gst_audio_test_src_do_seek), (gst_audio_test_src_check_get_range),
(gst_audio_test_src_create):
* gst/audiotestsrc/gstaudiotestsrc.h:
Define the default property values in the usual place.
Implement start/stop to reset values correctly.
Calculate the sample size only once when we negotiate.
Rename some values to make more sense.
Keep track of our byte range.
Add support for pull based scheduling. Disabled for now until we have
the whole stack working.
Set the BUFFER_OFFSET correctly.
Original commit message from CVS:
Based on a patch by: xavierb at gmail dot com
* gst/subparse/gstsubparse.c:
(gst_sub_parse_data_format_autodetect):
* tests/check/elements/subparse.c: (GST_START_TEST):
Make the detection of the used subtitle a bit less strict
for srt subtitles. Fixes bug #555607.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_pad_activate):
Remove bogus assert, the decodepad could have been created inside an
already existing group.
Original commit message from CVS:
2008-10-08 Andy Wingo <wingo@pobox.com>
* gst/playback/gstdecodebin2.c (expose_pad): Fix typo: unset
target instead of setting it.
(gst_decode_pad_activate, gst_decode_pad_unblock): This is now the
API for a decode pad. The bugfix is that we set the group in
activate(), not when the pad was created because it might be NULL
then.
(gst_decode_group_control_source_pad, gst_decode_group_expose):
Update to use the API.
Original commit message from CVS:
2008-10-08 Andy Wingo <wingo@pobox.com>
* gst/playback/gstdecodebin2.c (struct _GstDecodePad): Change to
be a subclass of GstGhostPad.
(analyze_new_pad): So, when emitting the signals that determine
how we do autoplugging, already create the ghost pad and use it as
the pad in the signal arguments. This allows applications to make
a connection between the pad passed in e.g. autoplug-continue, and
the pad passed in new-decoded-pad.
(connect_pad, expose_pad): Update to receive the ghosted decode
pad in the args, retargetting it as necessary if we have to plug
the target pad through a multiqueue.
(gst_decode_group_control_source_pad): Adapt to receive an
already-ghosted pad that just needs activation, blocking, and
drain notification.
(sort_end_pads): Adapt for decode pads actually being pads.
(gst_decode_group_expose): Adapt for decode pads actually being
pads. Rewrite the decode pad names so they appear in order. Adds a
new error case if we couldn't set the name.
(gst_decode_group_free, gst_decode_group_hide): Adapt cleanup
logic.
(gst_decode_pad_set_blocked, gst_decode_pad_add_drained_check):
New API for the decode pad, needed because we shouldn't do these
things inside gst_decode_pad_new(), but after.
(gst_decode_pad_new): Change to actually make the real pad, and
delay the blocking/drainage bits.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (audio_convert_convert):
Prevent overflows with big buffer when calculating the size of
the intermediate buffer by using gst_util_uint64_scale() instead of
plain arithmetics. Fixes bug #552801.
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c:
* gst/typefind/gsttypefindfunctions.c:
Recognise Kate subtitle streams (#550582).
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Typefind video/mj2 and image/jp2 ISO JPEG2000 mime types.
Fixes#550638.
Original commit message from CVS:
* configure.ac:
* gst/subparse/Makefile.am:
* gst/subparse/gstsubparse.c:
* gst/subparse/samiparse.c:
* tests/check/elements/subparse.c:
Rework last change, so that we build subparse, but just disable the
sami parse functionality, if we're configured to not use xml. In the
tests only the sami test is disabled now.
Original commit message from CVS:
Patch by: Jonathan Matthew <notverysmart gmail com>
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Add typefinder for PDF documents (which is nice to have, since it's a
common format, but also helps prevent false positives). Fixes#549814.
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (selector_blocked), (pad_added_cb),
(no_more_pads_cb):
Fix nasty race where multiple decodebins could start pushing data before
we manage to configure the sinks, resulting in not-linked errors in
typical RTSP streaming cases.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: DV typefinding. Remove
check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
Fixes#548065.
Original commit message from CVS:
2008-08-04 Andy Wingo <wingo@pobox.com>
* gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important
documentation fix.
Original commit message from CVS:
* gst/adder/gstadder.c:
Cleanup lots of empty lines that came from gst-indent going havoc
before I added the INDENT_ON/OFF marker some time agao.
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_getcaps),
(gst_video_test_src_create):
Discard buffers of the wrong size after renegotiation, this is perfectly
possible with things like capsfilter that could suggest caps changes
upstream without knowing the size of the buffer.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c:
Fix property doc markup (its not a signal).
* sys/xvimage/xvimagesink.c:
Add since tag for new proeprties (also add sice tags fro the last two
other additions).
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (celt_type_find),
(plugin_init):
Add simple typefinder for the CELT codec (www.celt-codec.org).
Original commit message from CVS:
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioquantize.c:
(gst_audio_quantize_setup_dither),
(gst_audio_quantize_free_dither):
* gst/audioconvert/gstfastrandom.h:
Implement a linear congruential generator as pseudo random number
generator for the dither noise. This is about 2 times faster than
using GLib's mersenne twister. Also this uses only integer math for
generating integers while GLib internally uses floating point math.
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c:
* gst/audiotestsrc/gstaudiotestsrc.h:
Add 'ticks', a 1/30 second sine wave pulse every second.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c:
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c:
* gst/playback/gststreamselector.c:
First stab at integrating DVD subpicture overlay into
playbin. Successfully plugs and plays, but the queues need
shrinking - 3 seconds of video is too much buffering.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (add_raw_queue):
And ref the pad before returning it again when linking to the queue
failed. Otherwise we will unref the pad twice later and things break.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (add_raw_queue):
If linking the raw pad with a queue fails, try it without a queue
instead of failing completely. This should never happen.
Original commit message from CVS:
Patch by: Evgeniy Stepanov <eugeni dot stepanov at gmail dot com>
* gst/playback/gstdecodebin.c: (add_raw_queue), (close_pad_link):
Add a queue after a demuxer if the demuxer outputs raw data. This was
done before only for non-raw data but is required in this case too.
Fixes bug #540215.
decodebin2 doesn't have this issue because all streams of a group
go through multiqueue.
Original commit message from CVS:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_get_unit_size):
Only set/get on the PAL8 format, ffmpegcolorspace doesn't support
it on other formats. Also adjust the unit size only for that format
to not include the palette. Fixes bug #540497.
Original commit message from CVS:
* gst/playback/gstplay-marshal.list:
* gst/playback/gstplaybin2.c:
Add get-video-pad, get-audio-pad, get-text-pad action signals to
playbin2. This allows the user to get to the selector's sinkpads, and
thus inspect a range of things - caps, tags, etc.
Original commit message from CVS:
* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
Fix a whole bunch of typos in comments and log statements.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
Ensure decodebin2 emits 'drained' signal once, and only once, when all
pads are drained.
Original commit message from CVS:
apparently it's an error to specify nc -l -p 3000 - though the short usage
does not make it very clear that you can drop the host arg with -l
Original commit message from CVS:
* gst/playback/gstplaybin2.c: (gst_play_bin_get_property),
(notify_source), (activate_group):
Implement the source property, emit notify when it changes in the
underlying uridecodebin.
Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/gdp/gstgdppay.c: (gst_gdp_pay_reset):
Fix a buffer memleak and remove a confusing and wrong debug output.
Fixes bug #538663.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c:
Disconnect signals from decodebins we created before we remove it from
playbin, to avoid crashes if the decodebin is eventually disposed after
the playbin itself (possible if the app takes a reference on the
decodebin).
Fixes#536521.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (aac_type_find),
(mp3_type_find), (musepack_type_find), (MULTIPART_MAX_HEADER_SIZE),
(mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
(h264_video_type_find), (mpeg_video_stream_type_find),
(dv_type_find), (mmsh_type_find):
Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't
copy caps for no good reason (this may be desirable to make it easier
to detect leaks, but then it should probably be done for all caps
in the typefinder somewhere).
Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader):
Don't set caps on the buffers that contain a copy of the buffer
including the caps of them resulting in an always increasing refcount
of the caps and insanely large caps. Instead include a buffer without
caps in the new caps. Fixes bug #536475.
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
Transform a given PAR to a range on the struct with the generic
height/width instead of the struct with the possibly restricted
height/width.
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
Prefer the given format if it contains something stricter than [1,MAX]
for height or width and only put a structure that requires rescaling
as second. This makes it possible to use videoscale in pipelines where
the source can actually produce the wanted height/width but usually
selects a different one from the requested.
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query_duration),
(gst_adder_query_latency):
When using gst_element_iterate_pads() one has to unref every pad
after usage.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(structure_has_fixed_channel_positions),
(gst_audio_convert_transform_caps):
Allow up to 11 positioned channels now that audioconvert can handle
this but add no default positions for > 8 channels.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add some unit tests for the above change: Test conversion of
11 positioned channels to stereo and the other way around, test
conversion of 15 unpositioned channels in different ways.
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c:
(gst_channel_mix_fill_one_other):
If mixing left or right to center (or the other way around) only take
the complete value if we don't already have the original position in
the source.
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions),
(gst_audio_set_structure_channel_positions_list),
(gst_audio_fixate_channel_positions):
Allow rear center together with rear left/right and other previously
conflicting channel positions. The reason why they weren't allowed
was the channel mixing implementation in audioconvert.
Also take this into account when fixing channel layouts.
Allow setting channel positions for 1/2 channels when using
gst_audio_set_structure_channel_position().
* gst/audioconvert/gstchannelmix.c:
(gst_channel_mix_fill_compatible), (gst_channel_mix_detect_pos),
(gst_channel_mix_fill_one_other), (gst_channel_mix_fill_others),
(gst_channel_mix_fill_special), (gst_channel_mix_fill_matrix):
Major rewrite of the channel mixing.
We now allow previously conflicting channel positions to appear
together (rear center and rear left/right for example).
Fixes bug #533817.
Rework the way channels are mixed together to take more possible
channel positions into account, properly mix from/to side channels
and don't assume that either center, left&right or nothing of a
specific position is available anymore.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Adjust unit tests with non-standard 1/2 channel layouts to the more
correct new behaviour.
Add a unit test for 5.1->Stereo downmixing.
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* gst/videorate/gstvideorate.c: (gst_video_rate_reset),
(gst_video_rate_flush_prev), (gst_video_rate_event),
(gst_video_rate_chain):
* gst/videorate/gstvideorate.h:
React (more) to NEWSEGMENT
Small adjustment in timestamp calculation to prevent mismatches
Fixes#435633.
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query_duration):
Correctly resync the iterator if gst_iterator_next() returns
GST_ITERATOR_RESYNC.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
(gst_decode_group_control_source_pad), (gst_decode_group_expose):
Check for NULL cases and log them, creating ghostpads can, for example,
fail when the pad returns wrong caps.
* gst/playback/gstplaybin2.c: (perform_eos):
When pushing out the EOS event, collect the return value and warn when
something failed.
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (DEFAULT_PROP_METHOD):
Change default scaling method from nearest-neighbour to bilinear.
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (parser_state_init),
(gst_sub_parse_format_autodetect), (handle_buffer):
* gst/subparse/gstsubparse.h:
* tests/check/elements/subparse.c: (test_tmplayer_style3b):
Limit duration to a maximum of five seconds for tmplayer format where
we can guess the duration only from the timestamp of the next line of
text. We don't want to show a text for eternities just because nothing
else is being said for a while.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/playback/gstdecodebin.c: (remove_fakesink):
Lock the fakesink before setting the state to NULL and removing it from
the bin so that a concurrent state change cannot interfere.
Fixes#534331.
Original commit message from CVS:
2008-05-21 Julien Moutte <julien@fluendo.com>
* gst/tcp/gstmultifdsink.c: (setup_dscp_client): Use IPPROTO_IP
instead of SOL_IP, works on more platforms.
* gst/typefind/gsttypefindfunctions.c: (aac_type_find): Fix printf
arguments.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_show_frame):
Some debug and comment fixes.
* tests/examples/dynamic/addstream.c: (main):
Fix , to ;
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to
other channel positions when source has SIDE channels and dest doesn't
or the other way around.
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c:
(gst_channel_mix_fill_normalize):
Prevent division by zero if the channel mix matrix contains only
zeroes.
Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/gdp/gstgdppay.c: (gst_gdp_pay_chain):
Close a buffer memory leak. Fixes bug #534071.
Original commit message from CVS:
* gst/adder/gstadder.c:
Adder also doesn't support audio/x-raw-int with width!=depth so don't
claim this on the pad template caps.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_channels):
Passthrough the channel positions if the number of output channels is
the same as the number of input channels, the input had a channel
layout and downstream requests no special one. We did this already for
> 2 channels but now it's also done for 1 channel. Fixes bug #533617.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_channels):
Correctly set the default channel positions when converting to 8
channels.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (aac_type_find):
Use data scan helper in aac typefinder and stop scanning
for headers when we've found a type. Also fix potential invalid
memory access when calculating the frame length.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (data_scan_ctx_ensure_data),
(mpeg_sys_is_valid_pack):
Don't modify scan context when we return FALSE in ensure_data, so
it's possible to continue scanning, and we don't end up with a NULL
data pointer and a positive size, which might bite us the next time
we're called. Small constification.
Original commit message from CVS:
* gst/adder/gstadder.c:
Adder doesn't support 24 bit samples so don't claim it supports them
in the pad template caps.
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
Revert previous change which made basetransform handle buffer_alloc
and which breaks things badly in the non-passthrough case since it
returned buffers with a different (ie. sometimes smaller) size than
the size requested.
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_setcaps),
(gst_adder_class_init):
Correctly declare the supported endianness on the pad templates
and check for correct endianness in the set caps function. Adder
only supports native endianness.
Also use gst_element_class_set_details_simple().
Original commit message from CVS:
Patch by: Hannes Bistry <hannesb at gmx dot de>
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
* gst/tcp/gsttcpserversink.c:
(gst_tcp_server_sink_handle_server_read),
(gst_tcp_server_sink_handle_wait), (gst_tcp_server_sink_init_send):
Fix regression in clientsrc because we did not add the fd to the poll
set anymore. Fixes#532364.
Do some cleanups here and there.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
* gst/playback/gstplay-marshal.list:
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
Use correct marshallers. GstCaps are a boxed type and no GObject
subclass.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
Don't do lots of 4-byte peeks, but use the 'new' data scan helper
for this instead; don't check if we've found enough markers after
each and every step, it's enough to do that only if we've actually
found a new marker.
Embed a G_UNLIKELY into the IS_MPEG_HEADER macro.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c:
(DATA_SCAN_CTX_CHUNK_SIZE), (DataScanCtx), (data_scan_ctx_advance),
(data_scan_ctx_ensure_data), (GST_MPEGVID_TYPEFIND_TRY_SYNC),
(mpeg_video_stream_type_find):
Move scan helper thingy to the beginning of the file so we can use
it in other typefind functions. Rename it to something more
generic. Also improve handling of things towards the end of the
typefind data: peek as much as we can if we know the size of the
data, rather than just min_size.
Original commit message from CVS:
Based on a patch by:
Björn Benderius <bjoern dot benderius at axis dot com>
* gst/ffmpegcolorspace/avcodec.h:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
(gst_ffmpegcsp_avpicture_fill):
* gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
* gst/ffmpegcolorspace/imgconvert_template.h:
Add conversions from/to NV12 and NV21 and conversions between those
two formats. Fixes bug #532166.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
Abort the h264 typefinding as soon as _peek() doesn't return anything,
which happens for example with files smaller than 128kb.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
Let audioresample use the buffer allocation of basetransform instead
of it's own stuff.
* tests/check/elements/audioresample.c: (alloc_only_48000),
(GST_START_TEST), (audioresample_suite):
Add unit test for the recent basetransform bugfix, where upstream
changes caps to something that can't be passed through anymore.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_transform_ip):
Return NOT_NEGOTIATED if we didn't set a process function yet for some
reason instead of crashing later. Might fix bug #509125.
Original commit message from CVS:
Based on a patch by: Tim-Philipp Müller <tim.muller at collabora co uk>
* gst/audioconvert/audioconvert.c: (audio_convert_prepare_context):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps),
(structure_has_fixed_channel_positions),
(gst_audio_convert_transform_caps):
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_matrix):
Add support for more than 8 channels and NONE channel layouts. For
more than 8 channels no channel conversion is supported yet, only
format conversions are supported. Fixes bug #398033.
* tests/check/elements/audioconvert.c: (verify_convert),
(GST_START_TEST), (audioconvert_suite):
Add some unit tests by Tim for checking the NONE channel layouts
and more than 8 channels and add some more unit tests for channel
conversions.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (connect_pad):
When autoplugging fails, set the element back to NULL before
unreffing it.
Original commit message from CVS:
* gst/subparse/samiparse.c: (handle_start_sync),
(end_sami_element), (characters_sami):
Remove trailing, leading and double whitespaces.
Correctly timestamp buffers and output the last buffer too.
* tests/check/elements/subparse.c: (GST_START_TEST),
(subparse_suite):
Add a simple unit test for SAMI parsing.
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian dot net>
* gst/subparse/samiparse.c: (handle_start_sync),
(start_sami_element), (end_sami_element), (characters_sami),
(sami_context_reset):
Only output characters inside the "sync" elements. There could be
other elements like "style" that have some content but should
not be printed. Fixes bug #467911.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (set_audio_mute),
(set_active_source):
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
(playbin_set_audio_mute):
Allow setting -1 as current-audio to mute the current audio stream,
similar to what is done for subtitles. Fixes bug #342294.
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (handle_buffer),
(gst_sub_parse_sink_event):
* gst/subparse/tmplayerparse.c: (tmplayer_process_buffer),
(tmplayer_parse_line):
Fix parsing of tmplayer subtitle variant where every single line contains
text and there isn't an empty line after each line to determine the
duration (#530962). Improve EOS handling for tmplayer subtitles a bit by
making sure that we push out the last line of text without a duration if
there's still text left in the buffer at the end.
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (feed_textbuf):
Fix detection of discontinuities based on the buffer offset (doesn't work
so well if no buffer offset is set) and also check for the DISCONT buffer
flag. This keeps the parser state from being reset after each buffer in
the unit test.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg_video_stream_type_find):
Further fine-tuning: don't absolutely require sequence or GOP headers
(as introduced in the previous commit), but adjust the typefind
probabilities returned accordingly if we don't see them. Also make sure
picture header and first slice are somewhat close to each other (which
is not perfect but still better than requiring a fixed offset or having
no limit at all).
Original commit message from CVS:
* gst/playback/test4.c:
* gst/playback/test5.c:
* gst/playback/test6.c:
* gst/playback/test7.c:
Also include config.h when relying on defines from it. Fixes the
build. Its been a please to serve :)
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c (paint_setup_NV12),
(paint_setup_NV21), (paint_hline_NV12_NV21):
Add support for NV12 and NV21 in videotestsrc
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c:
* gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
* gst/videoscale/vs_image.c: (vs_image_scale_nearest_RGBA),
(vs_image_scale_linear_RGBA), (vs_image_scale_nearest_RGB),
(vs_image_scale_linear_RGB), (vs_image_scale_nearest_YUYV),
(vs_image_scale_linear_YUYV), (vs_image_scale_nearest_UYVY),
(vs_image_scale_linear_UYVY), (vs_image_scale_nearest_Y),
(vs_image_scale_linear_Y), (vs_image_scale_nearest_RGB565),
(vs_image_scale_linear_RGB565), (vs_image_scale_nearest_RGB555),
(vs_image_scale_linear_RGB555):
Support 1x1 images as input and output as for example the BBC HQ new
streams have 1x1 GIFs in the playlists for some reason.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (free_pad_probe_for_element),
(try_to_link_1):
If we can't activate one of the decoders we plugged in (such as,
say, musepackdec) for some reason (it might not support push mode,
for example), remove any pad probes that close_pad_link() might
have set up. This makes sure we later don't try to remove a probe
for a pad that doesn't exist any longer, and avoids nast warnings
and probably other things too.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c:
(mpeg_video_stream_ctx_ensure_data), (mpeg_video_stream_type_find),
(plugin_init):
Rework mpeg video stream typefinding a bit more: make sure sequence,
GOP, picture and slice headers appear in the order they should and
that we've in fact at least had one of each; fix picture header
detection; decouple picture and slice header check - don't assume
they're at a fixed offset, there may be extra data in between. Also,
announce varying degrees of probability depending on what we found
exactly (multiple pictures, at least one picture, just sequence and
GOP headers). Finally, in _ensure_data(), take into account that we
might be typefinding smaller amounts of data, such as the first
buffer of a stream, so fall back to the minimum size needed as long
as that's available, instead of erroring out if there's less than
2kB of data. Fixes#526173. Conveniently also doesn't recognise the
fuzzed file from #399342 as valid.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (MpegVideoStreamCtx),
(mpeg_video_stream_ctx_advance), (mpeg_video_stream_ctx_ensure_data),
(mpeg_video_stream_type_find):
Refactor a bit: use context structure to track parsing offset and size of
available data and make the code a bit clearer. Fixes bad memory access
in #356937.
Original commit message from CVS:
* gst/playback/test4.c:
* gst/playback/test5.c:
* gst/playback/test6.c:
* gst/tcp/gstmultifdsink.c:
Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro
is defined.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
(plugin_init):
Add "mpp" and "mp+" as possible extensions for MusePack files.
Add typefinding for MusePack StreamVersion 8 files and include the
stream version in the caps.
Original commit message from CVS:
* docs/design/draft-keyframe-force.txt:
Fix typo.
* gst/playback/gstqueue2.c: (update_buffering),
(gst_queue_handle_src_query):
Set buffering mode in the messages.
Set buffering percent in the query.
* tests/examples/seek/seek.c: (update_fill), (msg_state_changed),
(do_stream_buffering), (do_download_buffering), (msg_buffering):
Do some more fancy things based on the buffering method in use.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (update_buffering),
(gst_queue_close_temp_location_file), (gst_queue_handle_src_query),
(gst_queue_src_checkgetrange_function):
Include extra buffering stats in the buffering message.
Implement BUFFERING query.
* gst/playback/gsturidecodebin.c: (do_async_start),
(do_async_done), (type_found), (setup_streaming), (setup_source),
(gst_uri_decode_bin_change_state):
Only add decodebin2 when the type is found in streaming mode.
Make uridecodebin async to PAUSED even when we don't have decodebin2
added yet.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
* gst/playback/gstplayback.c: (plugin_init):
* gst/volume/gstvolume.c: (plugin_init):
Work around missing bits of thread-safety on older GLibs some
more to avoid assertions when starting up multiple playbin
objects concurrently (see #512382).
