Turn _sink_event() into the collectpads event function and merge the logic from
the recently added gst_adder_event. Drop flush_start events as we allready
handle them on the src-pad side. Fixes#670850.
Now that we no longer support all methods for all formats, we
need to cater for that in the transform function: we can't
transform formats not supported by the currently-selected
mehod.
make check, folks. It's da bomb.
Only return LIKELY probability if we've seen an SPS, PPS and an
IDR slice nal, i.e. try harder to avoid false positives such
as with certain VC-1 files.
https://bugzilla.gnome.org/show_bug.cgi?id=668565
We need to call the default query handler of the proxy pad because only that one
will forward the query to the target pad in case of the allocation query.
After a PAUSED->READY change the sink pads are currently not set to
blocking state. When the element is set back to PAUSED, the change will
be done asynchronously, but as the _pad_blocked_cb() callback is now not
called, the state change never completes.
Fix that by setting the sink pads to blocking state on a PAUSED->READY
change, which ensures that the _pad_blocked_cb() is called when needed
on any future READY->PAUSED change. The sink pads are already put to
blocking state on NULL->READY change, so this behavior is consistent.
Fixes bug #668097.
In order to allow for proper functionality when a decoder only supports
one instance at a time (dsp), we must block the demuxer pads when they
get created if they are not part of the active group, preventing buffers
from being sent to the decoder (and initializing it through setcaps),
then after we switch to a new group, we unblock the demuxer pads for
the active groups. In the callback for the unblock, we prune the old
groups, making sure the previous decoder instance is destroyed before
we push a buffer to the new instance.
Since caps are no longer 'shared' between two pads (but forwarded from
source pad to sink pad) we end up with the first chain pad not having
specified caps (i.e. typefind:src).
This solves the issues by getting the pad's peer caps.
It is not optimal since it will (for most demuxers) return the pad
template caps, which might contain non-fixed caps (ex : with
qtdemux "video/quicktime; video/mj2; audio/x-m4a; application/x-3gp")
https://bugzilla.gnome.org/show_bug.cgi?id=667337
... to avoid unnecessary spurious errors (upon e.g. shutdown).
If a real error is applicable in this unusual circumstance (missing other pad),
other (STREAM_LOCK protected) call paths can take care of that.
We have removed things like protocol=gdp in the tcp elements
in favour of explicit gdppay/depay elements, so there's no need
to keep a public API and library for now. We can still add it
back later. Someone needs to think hard about 0.11 and gdp
anyway one of these days.
Make a new method to allocate a buffer + memory that takes the allocator and the
alignment as parameters. Provide a macro for the old method but prefer to use
the new method to encourage plugins to negotiate the allocator properly.
Improve GstSegment, rename some fields. The idea is to have the GstSegment
structure represent the timing structure of the buffers as they are generated by
the source or demuxer element.
gst_segment_set_seek() -> gst_segment_do_seek()
Rename the NEWSEGMENT event to SEGMENT.
Make parsing of the SEGMENT event into a GstSegment structure.
Pass a GstSegment structure when making a new SEGMENT event. This allows us to
pass the timing info directly to the next element. No accumulation is needed in
the receiving element, all the info is inside the element.
Remove gst_segment_set_newsegment(): This function as used to accumulate
segments received from upstream, which is now not needed anymore because the
segment event contains the complete timing information.
Hide the GstStructure of the event in the implementation specific part so that
we can change it.
Add methods to check and make the event writable.
Add a new method to get a writable GstStructure of the element.
Avoid directly accising the event structure.
So run-time bindings can introspect the names correctly (we abuse this
field as description field only in elements, not for public API
(where the description belongs into the gtk-doc chunk).
https://bugzilla.gnome.org/show_bug.cgi?id=629946
Adds that warning to configure.ac
Includes a tiny change of the GST_BOILERPLATE_FULL() macro:
The get_type() function is no longer declared before being defined.
https://bugzilla.gnome.org/show_bug.cgi?id=611692
Original commit message from CVS:
* libs/gst/dataprotocol/dataprotocol.c:
Don't write to the same region of memory as a uint64 and uint16
as this breaks strict aliasing rules and apparantly breaks on PPC
and s390. Thanks to Sjoerd Simons for analysing. Fixes bug #348114.
Original commit message from CVS:
* libs/gst/dataprotocol/dataprotocol.c:
(gst_dp_packet_from_event_1_0):
When calculating GDP body CRC, use the correct pointer.
Fixes part of #522401.
Original commit message from CVS:
2006-08-11 Andy Wingo <wingo@pobox.com>
* configure.ac:
* libs/gst/dataprotocol/dataprotocol.c: (gst_dp_packetizer_new):
* tests/check/libs/gdp.c: (gst_dp_suite): GST_DISABLE_DEPRECATED
is only for users of API that don't want to see deprecated
functions in the headers; people that want to compile out
deprecated code should pass -DGST_REMOVE_DEPRECATED into the
CFLAGS. Fixes the build of multifdsink, or will soon..
Original commit message from CVS:
* libs/gst/dataprotocol/dataprotocol.c:
(gst_dp_header_from_buffer_any), (gst_dp_packet_from_caps_any),
(gst_dp_crc), (gst_dp_header_payload_length),
(gst_dp_header_payload_type), (gst_dp_packet_from_event),
(gst_dp_packet_from_event_1_0), (gst_dp_buffer_from_header),
(gst_dp_caps_from_packet), (gst_dp_event_from_packet_0_2),
(gst_dp_event_from_packet), (gst_dp_validate_header),
(gst_dp_validate_payload):
Make debug category static
Constify the crc table.
Do some more arg checking in public functions.
Fix some docs and do some small cleanups.
* tests/check/libs/gdp.c: (GST_START_TEST), (gst_dp_suite):
Add some more checks to see if GDP deals with bogus input.
Original commit message from CVS:
* libs/gst/dataprotocol/dataprotocol.c:
(gst_dp_event_from_packet_1_0):
Fixes#347337: failure to deserialize event packets with
empty payload (only event type)
Original commit message from CVS:
* docs/README:
* docs/images/gdp-header.svg:
add a gdp image
* docs/libs/Makefile.am:
* docs/libs/gdp-header.png:
* libs/gst/dataprotocol/dataprotocol.c:
add it to the API docs
* docs/manual/intro-motivation.xml:
fix typo
Original commit message from CVS:
* libs/gst/dataprotocol/dataprotocol.c:
(gst_dp_header_from_buffer), (gst_dp_packet_from_caps),
(gst_dp_packet_from_event):
factor out CRC code
Original commit message from CVS:
* libs/gst/dataprotocol/dataprotocol.c:
(gst_dp_header_from_buffer), (gst_dp_packet_from_caps),
(gst_dp_packet_from_event):
factor out some common header init code
Original commit message from CVS:
* docs/libs/gstreamer-libs-sections.txt:
* docs/libs/tmpl/gstdataprotocol.sgml:
* libs/gst/dataprotocol/dataprotocol.c: (gst_dp_crc):
* libs/gst/dataprotocol/dataprotocol.h:
API: make gst_dp_crc() public
Original commit message from CVS:
* libs/gst/dataprotocol/dataprotocol.c: (gst_dp_packet_from_event),
(gst_dp_event_from_packet):
Fixes in reading/writing events over GDP (not currently used?) -
dereferencing NULL events for unknown/invalid event types, memory
leak, and change g_warning to GST_WARNING.
Original commit message from CVS:
* libs/gst/dataprotocol/dataprotocol.c:
Fix docs for dataprocotol to not get the return types completely
wrong for a few functions.
Original commit message from CVS:
2005-10-13 Andy Wingo <wingo@pobox.com>
* libs/gst/dataprotocol/dataprotocol.c (gst_dp_packet_from_caps):
Fix Timmeke Waymans bug.
(gst_dp_caps_from_packet): Make sure we pass a NUL-terminated
string of the proper length to gst_caps_from_string. There's a
potential for, before this fix, that this could cause someone
connecting over the network to cause a segfault if the payload is
not NUL-terminated.
Original commit message from CVS:
* libs/gst/dataprotocol/dataprotocol.c:
(gst_dp_header_from_buffer), (gst_dp_packet_from_caps),
(gst_dp_packet_from_event):
* libs/gst/dataprotocol/dataprotocol.h:
* libs/gst/dataprotocol/dp-private.h:
It's about time we bump the version number.
Since event types don't fit in the guint8 anymore describing
the payload type, make payload type 16 bits wide.
Original commit message from CVS:
* libs/gst/dataprotocol/dataprotocol.c: (gst_dp_packet_from_event),
(gst_dp_event_from_packet):
Fix serialization of seek events.
Original commit message from CVS:
Next big merge.
Added GstBus for mainloop integration.
Added GstMessage for sending notifications on the bus.
Added GstTask as an abstraction for pipeline entry points.
Removed GstThread.
Removed Schedulers.
Simplified GstQueue for multithreaded core.
Made _link threadsafe, removed old capsnego.
Added STREAM_LOCK and PREROLL_LOCK in GstPad.
Added pad blocking functions.
Reworked scheduling functions in GstPad to prepare for
scheduling updates soon.
Moved events out of data stream.
Simplified GstEvent types.
Added return values to push/pull.
Removed clocking from GstElement.
Added prototypes for state change function for next merge.
Removed iterate from bins and state change management.
Fixed some elements, disabled others for now.
Fixed -inspect and -launch.
Added check for GstBus.
Original commit message from CVS:
First THREADED backport attempt, focusing on adding locks and
making sure the API is threadsafe. Needs more work. More docs
follow this week.
Original commit message from CVS:
2005-02-18 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* libs/gst/dataprotocol/dataprotocol.c: (gst_dp_dump_byte_array):
Allocate the 1 byte more memory that was forgotten!!!!!
Flesh out the video filter base class. Make it parse the input and output caps
and turn them into GstVideoInfo. Map buffers as video frames and pass them to
the transform functions.
This allows us to also implement the propose and decide_allocation vmethods.
Implement the transform size method as well.
Update subclasses with the new improvements.
With the new video bufferpool we can now implement the propose_allocation
vmethod on some video filter elements so that we can also use video metadata and
bufferpools when not operating in passthrough mode.
GstCollectPads2 locking was changed from GstCollectPads to use
the stream lock instead of the object lock for those cases, so
change it so here as well to match.
https://bugzilla.gnome.org/show_bug.cgi?id=666379
... to also properly indicate chain's endpad if no elements are in the
chain (due to the endpad being a raw demuxer pad, or one setup without
decoders since uridecodebin or higher up decided not to need those).
Previously we always used textoverlay for rendering the output of
a parser, now the same code as for the renderers is used and the
element with the highest rank is used.
Fixes bug #663822.
We added the utf typefinder because the mp3 typefinder was a tad
overzealous when it came to typefinding things as mp3, and replaced
it with even more overzealous utf16/32 typefinders.
Fixes unit test.
This reverts commit bd539753eb.
Adding the supported metadata to the query does nothing at this stage. Proposing
allocation parameters and supported metadata for upstream should use the
propose_allocation vmethod.
Add private replacements for deprecated functions such as
g_mutex_new(), g_mutex_free(), g_cond_new() etc., mostly
to avoid the deprecation warnings. We'll change these
over to the new API once we depend on glib >= 2.32.
Replace g_thread_create() with g_thread_try_new().
Make appsink return a GstSample. Remove the pull_buffer_list method because it
is not very useful anymore.
Pass GstSample to the conversion function.
Update playbin2 and examples
The output size of a buffer does not depend on the input size but simply on the
caps of the output buffers. Don't let the base implementation deal with
unit_sizes, because input buffers might not be a multiple of that when they have
padding or non-default strides. instead, implement a transform size function
that simply calculate the natural size of an output buffer based on the caps.
Doing dynamic pipelines is hard in 0.10. As we don't have the sticky events in
0.10 and sending such events in special elements like adder and tee was outvoted
on last attempt, be graceful to the misbehaviour instead.
This happens when the internal elements are added before any NEWSEGMENT
event arrived and in that case we shouldn't send a NEWSEGMENT event
to the internal elements at all. They will get the NEWSEGMENT event
from upstream later.
If the sink supports raw audio/video, we first check
if the decoder could output any raw audio/video format
and assume it is compatible with the sink then. We don't
do a complete compatibility check here if converters
are plugged between the decoder and the sink because
the converters will convert between raw formats and
even if the decoder format is not supported by the decoder
a converter will convert it.
We assume here that the converters can convert between
any raw format.
Fixes bug #665120.
fix build errors:
gsttypefindfunctions.c:248:25: error: 'low' may be used uninitialized in this function [-Werror=uninitialized]
gsttypefindfunctions.c:239:24: error: 'high' may be used uninitialized in this function [-Werror=uninitialized]
Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net>
audioresample is derived from GstBaseTransform, and one of
GstBaseTransform's traits is that if the derived element does not
produce an output buffer from some input buffer then the first output
buffer after that gets flaged as a discontinuity, whether or not the
buffer actually is discontinuous from the output buffer that preceded
it. When downsampling, the audioresample element requires more than
one input sample for each output sample, and if the ratio of input to
output sample rates is high enough and the input buffers short enough
it can come to pass that the resampler does not receive enough samples
on its input to produce any output. Currently the resampler returns
GST_BASE_TRANSFORM_FLOW_DROPPED from the transform() method in this case,
causing the next buffer to be flagged as a discontinuity. If subsequent
elements in the pipeline reset themselves on disconts, this can cause
clicks and other undesireable behaviour.
Fixes bug #665004.
After preroll the multiqueue limits are still set to the preroll
limits if use-buffering is set to TRUE. In that case we only want
time limits on the multiqueue if upstream is seekable.
Such streams were detected as seekable, as the query on the typefind
element was testing the m3u8 file listing the actual streams, and
not going through the demuxer(s).
We now check for seekability for each multiqueue following a demuxer,
so the query will flow through the elements which might prevent seeking.
https://bugzilla.gnome.org/show_bug.cgi?id=647769
API: GstVideoRate:force-fps
Changing the framerate during playback is not possible
with a capsfilter downstream if upstream is not using
gst_pad_alloc_buffer(). In that case there's no way in
0.10 to signal to videorate that the preferred framerate
has changed.
This new property will force the output framerate to
a specific value and can be changed during playback.
The ghostpad acceptcaps functions are not valid in this case because
we don't only accept the caps accepted by the target but could also
insert converters. Fixes bug #663892.
This allows us to easily get ahold of all pads on a stream-topology message, including
pre-decoder ones, while "pad" only gives us access to the raw pads (as used by discoverer).
Set up targets on READY->PAUSED state change to passthrough by
default. This prevents the targets from being unset on the
first run, while the 'raw' variable would mean that some
target is set.
The identity element should be handled by the GstBin's cleanup,
removing it on the remove_elements function might remove it
too soon, as this function can be called directly from playsink
The playsink was nastily poking a boolean in the structure.
Make those booleans properties, so we are told when they change,
and rebuild the conversion bin when they do.
Some cleanup to go with it too.
https://bugzilla.gnome.org/show_bug.cgi?id=661262
ie, audio/x-raw- for audio, video/x-raw- for video.
Add a trailing - to be more specific. I doubt there's anything
like audio/x-rawhide or something, but you never know.
https://bugzilla.gnome.org/show_bug.cgi?id=661262
The code was doing counterintuitive rewiring of pads when the
bin did not contain any elements. We now add an identity element
in that case, which makes it simpler, and should fix the AC3
passthrough mode when using pulseaudio (but I don't see the bug
here so can't test).
https://bugzilla.gnome.org/show_bug.cgi?id=661262
This is made possible by filtering errors. This is required to let
harware accelerated element query the video context. The video context
is used to determine if the HW is capable, and thus if the element is
supported or not.
Fixes bug #662330.
If the pad block never happens because there is no data flow at all, the
callback is never fired and the reference is never released. This causes a
reference cycle between the pad and element, so valgrind is not very vocal
about it (memory is still reachable).
The bins' getcaps was bypassing the inner elements, and thus
failing to account for the caps transformations they allow,
which caused YUV video pipelines to fail with ximagesink, which
does not support YUV, even though the convenience bin includes
a colorspace converter for just this purpose.
https://bugzilla.gnome.org/show_bug.cgi?id=660816
The new code was checking for a prefix, and would find video/
first. Check in two passes, first checking for a perfect match,
and falling back to a prefix check if nothing was found.
https://bugzilla.gnome.org/show_bug.cgi?id=657261
Re-enable parsers in encodebin to allow more passthrough scenarios
to work. Specially the ones that require changing 'stream formats'.
i.e. h264 in mkv to mpegts.
The fact that a decoder is not compatible with the fixed sink
is currently happenning in the case where we have hardware accelerated
video decoders on the system (especially vaapi elements that are actually plugged),
and the user is providing a sink that doesn't support the surface.
A simple example that shows how it used to crash on a system where gstreamer-vaapi
is installed:
gst-launch playbin2 video-sink=xvimagesink uri=/codec/supported/by/vaapi
What we are now doing in this case, is avoid using the accelerated
decoder and plug a "normal" decoder instead (if avalaible).
This commit doesn't handle the case where we have hardware accelerated
demuxing.
The condition is if the muxer doesn't have tag setter *and* isn't
a formatter itself. Any of those two conditions makes the muxer
good enough to not need a formatter.
gstsubtitleoverlay.c: In function 'gst_subtitle_overlay_video_sink_event':
gstsubtitleoverlay.c:1736:22: error: 'target' may be used uninitialized in this function
There's no code whatsoever that uses these macros. If anyone
ever feels the need to resurrect them, we should add them to
gstutils.h in core or libgstaudio or so.
In various use-case you want to dynamically change the framerate (e.g.
live streams where the available network bandwidth changes). Doing this
via capsfilters in the pipeline tends to be very cumbersome and racy,
using this property instead makes it very painless.
With unfixed caps we can't reliably decide if the final caps
are going to be "raw" (e.g. supported by a sink) or not.
We will get here again later when the caps are fixed.
If subdrained isn't initialized to FALSE then a chain might think
that its group is drained when in fact it's not and this can cause
a switch too early or even cause a deadlock.
This reverts commit b0b4e286c8.
We agreed that the previous (pre-.35) behaviour is broken and a bug and the
current behaviour is correct, deterministic and allows the application to
handle stuff properly while the old behaviour can't be handled properly by
applications and just worked in some applications by luck.
The solution to the problem that was solved by relying on the old, broken
behaviour would be, to make decodebin2/playbin2 more aware of decoders and
improve the autoplugging of decoders by considering the caps supported by the
sink instead of just using something with the highest rank.
See bug #656923.
Fixes regression since 0.10.33 where sinks that can cope with non raw
caps or custom caps are not autoplugged if there's a sink configured
with the properties video-sink and audio-sink which cannot handle
the stream. This change checks for compatibility on the configured one
and use it if success. Otherwhise it tries with the found factories.
This reverts commit a22faad18a. Instead
of disabling subtitles completelly when video stream have custom caps,
just let the sutbtileoverlay cope with them as now it's able to.
Implement handling of non raw video streams by avoiding colorspace
elements and autoplugging a compatible renderer if available. Fallback
to passthrough if no compatible renderer is found.
Only log in debug log for now, since the check is a bit
half-hearted, its purpose is mostly to make sure people
use gst_filename_to_uri() or g_filename_to_uri().
https://bugzilla.gnome.org/show_bug.cgi?id=654673
Note that there is already a AMF detection for a different
magic, I'm not sure if that's a different format with the
same initials or not. AMF is used for a few different formats
(including video), so...
This fixes playbin2 playing Asylum modules.
https://bugzilla.gnome.org/show_bug.cgi?id=658514
This patch prevents timestamp like "1 1:00:00", which would have been seen
as hour 101 by our parser, and allow single digit hour, minute and seconds
as it's already supported by the parser, and also by other implementation
like in mplayer. This fixes bug 657872.
https://bugzilla.gnome.org/show_bug.cgi?id=657872
g_value_get_object() does not give us our own ref.
Fixes "Trying to dispose object "flacparse", but it still has a parent "registry0".
You need to let the parent manage the object instead of unreffing the object directly."
and similar warnings.
https://bugzilla.gnome.org/show_bug.cgi?id=658416
This is done by adding a capsfilter after every parser/converter that contains
all possible caps supported by downstream elements. A capsfilter is necessary
here because the decoder is only selected after the parser selected a format
and the parser can't know what downstream would support otherwise.
Remove the _ in front of the endianness prefix.
Remove the _3 postfix for the 24 bits formats.
Add a _32 postfix after the formats that occupy extra space beyond their
natural size.
The result is that the GST_AUDIO_NE() macro can simply append the endianness
after all formats and that we only specify a different sample width when it is
different from the natural size of the sample. This makes things more consistent
and follows the pulseaudio conventions instead of the alsa ones.
Sort muxers based on their caps and ranking before iterating to
find one that fits the profile.
Sorting is done by putting the elements that have a pad template
that can produce the exact caps that is on the profile. For example:
when asking for "video/quicktime, variant=iso", muxers that
have this exact caps on their pad templates will be put first on
the list than ones that have only "video/quicktime".
https://bugzilla.gnome.org/show_bug.cgi?id=651496
This reverts commit 105814e2c7.
The general consensus seems to be that we should revert this for
now. If such behaviour is desired, we should probably enable it
via a flag. And maybe use the scaletempo plugin instead.
Adds a Lanczos-derived scaling method, which is rather slow, but very
high quality. Adds a few properties that can be used to tune various
scaling properties: sharpness, sharpen, envelope, dither. Not currently
Orcified, but was designed with that in mind.
The average_period_set variable can be accessed in different threads, so
always lock it when reading. Furthermore when switching to averaging
mode we should make sure we don't have cached buffers that aren't used
in that mode. And any modeswitch will cause the latency to change, so we
should post a NewLatency message
Make enums for the chroma siting for easier use in the videoinfo.
Make enums for the color range, color matrix, transfer function and the
color primaries. Add these values to the video info structure in a Colorimetry
structure. These values define the exact colors and are needed to perform
correct colorspace conversion. Use a couple of predefined colorimetry specs
because in practice only a few combinations are in use.
Add view_id to the video frames to identify the view this frame represents in
multiview video.
Remove old gst_video_parse_caps_framerate, use the videoinfo for this.
Port elements to new colorimetry info.
Remove deprecated colorspace property from videotestsrc.
Rework the audio caps similar to the video caps. Remove
width/depth/endianness/signed fields and replace with a simple string
format and media type audio/x-raw.
Create a GstAudioInfo and some helper methods to parse caps.
Remove duplicate code from the ringbuffer and replace with audio info.
Use AudioInfo in the base audio filter class.
Port elements to new API.
Instead of just assuming all pads are created at the same time,
remember which ones are actually new (via ->pending_blocked_pads).
This allows the following use-case to properly work:
* Upstream starts with audio-only
* Only that pad gets data, blocks and a real audio sink is created
* Upstream laters adds a video stream
* A new pad is requested, blocks and reconfiguration kicks in in
order to add a new real video sink
Similar meaning same layer, same bitrate, and same number of channels
This fixes misdetection of (some MP3 files that have zero padding
between the ID3 tag and the MP3 stream) as H.264 video.
https://bugzilla.gnome.org/show_bug.cgi?id=656018
As encodebin doesn't connect to the queue signals, it can set
queues to silent mode to make queue not emit them.
Check https://bugzilla.gnome.org/show_bug.cgi?id=621299 for
more info on queue's silent property.
Use atomic ops on pending flags. Rename the segment_pending to
new_segment_pending. Set new_segment_pending not when we received seek, but
when we received the first upstream new_segment.
When we don't have specific {audio|video|text}-sink properties, don't
set them on playsink when reconfiguring.
If we do that, we end up setting the previous configured sink to
GST_STATE_NULL resulting in any potentially pending push being returned
with GST_FLOW_WRONG_STATE which will cause the upstream elements to
silently stop.
https://bugzilla.gnome.org/show_bug.cgi?id=655279
When we have a multi-stream (i.e. audio and video) input and the demuxer
adds/removes pads for a new stream (common in a mpeg-ts stream when the
program stream mapping is updated), the algorithm for EOS handling was
previously wrong (it would only drop the EOS of the *last* pad but would
let the EOS on the other pads go through).
The logic has only been changed a tiny bit for EOS handling resulting in:
* If there is no next group, let the EOS go through
* If there is a next group, but not all pads are drained in the active
group, drop the EOS event
* If there is a next group and all pads are drained, then the ghostpads
will be removed and the EOS event will be dropped automatically.
This allows us to make parsers accept both parsed and unparsed input
without decodebin plugging them in a loop until things blow up, ie.
without affecting applications that still use the old playbin or the
old decodebin.
(Making parsers accept parsed input is useful for later when we want
to use parsers to convert the stream-format into something the decoder
can handle. It's also much more convenient for application authors
who can plug parsers unconditionally in transcoding pipelines, for
example).
Make a new GstVideoFormatinfo structure that contains the specific information
related to a format such as the number of planes, components, subsampling,
pixel stride etc. The result is that we are now able to introduce the concept of
components again in the API.
Use tables to specify the formats and its properties.
Use macros to get information about the video format description.
Move code to set strides, offsets and size into one function.
Remove methods that are not handled with the structures.
Add methods to retrieve pointers and strides to the components in the video.
Add a flags property and two flags to allow one to disable the
conversion elements within encodebin. Doing so insists that the
uncompressed input to encodebin for the appropriate stream type is
sufficient to meet the caps requirements of the encoders, muxers and
encodebin target.
This is mostly beneficial to bypass slow caps negotiations in the
conversion elements.
Caps returned from gst_pad_peer_get_caps_reffed () may not be writable.
If they are not is should cause an assertion in gst_caps_merge (),
however, sometimes assertions are disabled in binary builds of -base and
it's safer to just be sure the caps are writable. Also, check that the
reffed caps pointer is not NULL.
The length check isn't sufficient, an source might
report the correct length, but then still fail to
read the requested number of bytes for some reason.
https://bugzilla.gnome.org/show_bug.cgi?id=652642
Remove the GstVideoPlane structure and move the fields directly into the
GstVideoInfo structure. This makes things a little easier to read and also makes
it more likely that we can pass the stride array to external libraries.
Remove the android/ top dir
Fixe the Makefile.am to be androgenized
To build gstreamer for android we are now using androgenizer which generates the needed Android.mk files.
Androgenizer can be found here: http://git.collabora.co.uk/?p=user/derek/androgenizer.git
Work in progress. Colorspace handles most format conversion using
3-stage getline/matrix/putline process using an AYUV or ARGB
intermediate, with most functions handled by Orc. There is also
a table of single-pass conversions, all handled by Orc. The plan
is to add optional stages for various chroma upsampling/downsampling
algorithms, dithering, and float/int16 intermediates, and then have
Orc create multi-stage functions at runtime.
Original commit message from CVS:
* ext/hermes/gsthermescolorspace.c: (plugin_init): decrease rank
by 2 to not interfere with other colorspaces.
* ext/pango/gsttextoverlay.c: (plugin_init): change rank to NONE
* gst/colorspace/gstcolorspace.c: (plugin_init): decrease rank by
one to not interfere with ffmpeg_colorspace.
Original commit message from CVS:
* configure.ac: the Hermes library controls hermescolorspace, not
colorspace.
* ext/mpeg2dec/gstmpeg2dec.c: (gst_mpeg2dec_base_init),
(gst_mpeg2dec_init): minor pet peeve: disable code with #ifdef,
not /* */
* ext/sdl/sdlvideosink.c: Change XID to unsigned long.
* ext/sdl/sdlvideosink.h: ditto.
* gst/colorspace/gstcolorspace.c: Fix old comments about Hermes
Original commit message from CVS:
* gst-libs/gst/audio/Makefile.am:
Add gstaudiofiltertemplate.c and building of gstaudiofilterexample.c
from the template.
* gst-libs/gst/audio/gstaudiofilter.c:
* gst-libs/gst/audio/gstaudiofilter.h:
Add bytes_per_sample and size and n_samples calculation.
* gst-libs/gst/audio/gstaudiofilterexample.c:
Remove, now autogenerated.
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
Moved from gstaudiofilterexample, object name changed, code added
so that it actually works.
* gst-libs/gst/audio/make_filter:
Script to build an audiofilter subclass from the template.
* gst/colorspace/Makefile.am:
* gst/colorspace/yuv2yuv.c:
Remove file, since it's GPL, and we don't use it.
Original commit message from CVS:
2004-01-15 Ronald Bultje <rbultje@ronald.bitfreak.net>
* gst/colorspace/gstcolorspace.c:
* gst/colorspace/yuv2yuv.c: (gst_colorspace_yuy2_to_i420),
(gst_colorspace_i420_to_yv12):
Fix compiling... Didn't test if it actually works.
Original commit message from CVS:
* configure.ac:
* gst/colorspace/Makefile.am:
* gst/colorspace/gstcolorspace.c:
* gst/colorspace/gstcolorspace.h:
* gst/colorspace/yuv2rgb.c:
* gst/colorspace/yuv2rgb.h:
Duplicate the ext/hermes colorspace plugin, and remove Hermes
code and GPL code. Fix for new caps negotiation. Rewrite
much of the format handling code, and some of the conversion
code. Basically, rewrote almost everything. This element
handles I420, YV12 to RGB conversions.
* ext/hermes/Makefile.am:
* ext/hermes/gsthermescolorspace.c:
Rename colorspace to hermescolorspace. Fix negotiation issues.
Remove non-Hermes related code. This element handles lots of
RGB to RGB conversions, but no YUV.
* ext/hermes/gstcolorspace.c:
* ext/hermes/gstcolorspace.h:
* ext/hermes/rgb2yuv.c:
* ext/hermes/yuv2rgb.c:
* ext/hermes/yuv2rgb.h:
* ext/hermes/yuv2rgb_mmx16.s:
* ext/hermes/yuv2yuv.c:
* ext/hermes/yuv2yuv.h:
Remove old code.
The old v4l interface has been deprecated for years and even
been removed from the kernel headers. If anyone still needs
this plugin, they can resurrect it in gst-plugins-bad, there's
no reason for it to be in -base.
If the second and next caps structures are a subset of the already existing
transformed caps we can safely skip them because we would transform them to
the same caps again.
If the second and next caps structures are a subset of the already existing
transformed caps we can safely skip them because we would transform them to
the same caps again.
If the second and next caps structures are a subset of the already existing
transformed caps we can safely skip them because we would transform them to
the same caps again.
If the second and next caps structures are a subset of the already existing
transformed caps we can safely skip them because we would transform them to
the same caps again.
This makes gst_pad_get_caps() on an audiotestsrc ! audioconvert !
audioconvert ! audioconvert ! fakesink pipeline about 1.7 times faster.
This is especially needed when switching between a non-sparse and sparse
video stream, see bug #537382. It also lowers the time needed for switching
between streams a bit.
Previously we checked mute_csource to determine wheter we need to premultiply
volumes and mute values. That fails as we unrefs mute_csource and set it to
NULL after. Use an extra flag instead.
make_lossless_changes() returns the same structure that we're passing (probably
to enable chaining). Instead of reusing s and making it point to s2 as well,
keep using s2. Drop the assignment which in the 2nd case is a dead one anyway.
Autoplug formatters for streams if a formatter with secondary or
higher rank is found. Formatters are autoplugged when there is no
muxer or when the muxer doesn't implement the tagsetter interface.
Currently only the first formatter found is plugged, this might
help in lots of cases, but it doesn't solve the
'lamemp3 ! xingmux ! id3mux'
case.
https://bugzilla.gnome.org/show_bug.cgi?id=649841
In particular, in audio only cases whose (estimated) metadata provides bitrate
information, the buffer-size based on such bitrate (and buffer-duration)
will be much more reasonable than queue2 default buffer-size.
For streams at low bitrates we need to set a limit in time because the limit
in bytes might not reached too late, sometimes more than 30 seconds.
This limit can only be set if upstream is seekable (see #584104)
Closes#647769
These reconfigure based on the caps and plugin in converters if
necessary. This also makes switching between compressed and raw
streams work flawlessly without loosing the states of any element
somewhere or having running time problems.
Before playbin2 would use different selectors for raw audio and
compressed audio (and the same for video) and used different
pads from playsink. This made the involved logic much more
complex and was not implemented completely in playsink, which
made it impossible to support files with a compressed and
uncompressed stream that is support by the sink.
playbin2 handles raw/non-raw streams the same now and the
decision is left to playsink, which now can also handle
caps changes from raw to non-raw and the other way around.
Fixes bug #632788.
Fixes#648548. Orc generates bad code for
gst_videoscale_orc_resample_merge_bilinear_u32, so we'll use the
slightly slower two-stage process. I'd fix Orc, but it's hard to
get excited about fixing a feature that I'm planning to deprecate
and replace.
Remove the android/ top dir
Fixe the Makefile.am to be androgenized
To build gstreamer for android we are now using androgenizer which generates the needed Android.mk files.
Androgenizer can be found here: http://git.collabora.co.uk/?p=user/derek/androgenizer.git
Remove the android/ top dir
Fixe the Makefile.am to be androgenized
To build gstreamer for android we are now using androgenizer which generates the
needed Android.mk files.
Androgenizer can be found here:
http://git.collabora.co.uk/?p=user/derek/androgenizer.git
We should keep playlist/m3u8 available for normal m3u8 playlists,
which we we'll likely support some day. Also, we probably don't
want this handled like other playlists, so application/* seems
more appropriate in this case, even if it's really just a playlist.
In addition to ensuring that an element we want to select in
autoplug-select can enter the READY state, we also now check if it can
accept the caps we wish to plug it for. This is handy for sinks that
need to perform a probe to figure out whether they can actually handle a
given format.
When fixating caps, from_par should always be initialized
with a fixed value.
In case the fixation is from src to sink pad it was setting
the from par (srcpad par) to a fraction range, this patch initializes
it to 1/1, based on the assumption that missing PAR is 1/1.
https://bugzilla.gnome.org/show_bug.cgi?id=641952
Post better error messages in case typefind/decodebin2 are missing or
could not be loaded for some reason (e.g. because they inadvertently
got blacklisted).
https://bugzilla.gnome.org/show_bug.cgi?id=644892
In NULL/READY, we should be able to switch profiles on encodebin,
this patch makes it tear down old profiles when new ones are set
if in NULL/READY states
https://bugzilla.gnome.org/show_bug.cgi?id=644416
Clients are usually disconnected in the streaming thread if their inactivity
is bigger than the timeout. If no new buffers are to be rendered in the sink,
these clients will never be disconnected and for that reason it should be
handled in the select() loop too.
Clients are usually disconnected in the streaming thread if their inactivity
is bigger than the timeout. If no new buffers are to be rendered in the sink,
these clients will never be disconnected and for that reason it should be
handled in the select() loop too.
Parsers are the only element class that are not changing the data and
could lead to an infinite loop. Other element classes like demuxers,
e.g. id3demux, can be used multiple times in a row and sometimes are.
Previously we only checked against the raw caps but we should also
check against the return value of autoplug-continue. Additionally fix
a thread-safety issue with accessing the raw caps.
Add "source-setup" signal for convenience and discoverability. No need
to figure out "notify::source", look up the notify callback signature,
then do an g_object_get() to get the source element..
https://bugzilla.gnome.org/show_bug.cgi?id=626152
As a result, pipelines that contain multiple instances of audiotestsrc
with the 'wave' property set to 'white-noise', 'pink-noise', or
'gaussian-noise' will run much faster, since they won't be competing
for access to the global, lock-protected instance of GRand.
Fixes bug #642720.
...instead of copying the array. Returning NULL will result
in the original factories array to be used and prevents a useless
array copy in most use cases.
...instead of copying the array. Returning NULL will result
in the original factories array to be used and prevents a useless
array copy in most use cases.
Add notes about the behaviour if multiple signal handlers are connected.
For most autoplug-* signals only the first signal handler will ever
be invoked.
Also add to the autoplug-sort docs that the signal handler can return NULL
to specify that the order should change and other handlers get the chance
to sort the array.
This lock is taken when activating a group, which could result in
calling the autoplug-continue callback, which also needs this lock
to access the sinks.
See bug #642174.
Don't build merge the caps of all sinks but check them one-by-one
until one supports the caps. Also get reffed caps from the sinkpads
instead of a writable copy and add debug output if a sink claims to
support ANY caps.
The outgoing buffer timestamp is calculated by scaling an output buffer
count by the src pad frame rate caps. If these caps change, we need to
reset the count and work from a new base timestamp. The new output
buffer timestamp is then the count scaled by the new caps values added
onto the base timestamp.
with i686-apple-darwin10-gcc-4.2.1:
encoding-profile.h:134: warning: type qualifiers ignored on function return type
encoding-profile.c:240: warning: type qualifiers ignored on function return type
gstencodebin.c: In function 'next_unused_stream_profile':
gstencodebin.c:454: warning: format '%d' expects type 'int', but argument 8 has type 'GType'
gstencodebin.c:464: warning: format '%d' expects type 'int', but argument 8 has type 'GType'
Since we calculate timestamps by:
timestamp = t0 + (out samples) / (out rate)
and durations by:
duration = ((out samples) + (processed samples)) / (out rate) - timestamp
if t0 is nonzero, this would simplify to
duration = t0 + (processed samples) / (out rate).
This duration is too large by the amount t0. We should have done:
duration = t0 + ((out samples) + (processed samples)) / (out rate) - timestamp
so that
duration = (processed samples) / (out rate).
Frame size is given in words; it is already multiplied by two where
needed, so the left shift is superfluous. This extra multiplication
caused the code to inspect the third packet instead of the second,
which would fail for files where the second packet has a size
different from the first.
Some things aren't quite right yet and cause problems (0-sized buffers
with PREROLL flag set cause crashes in elements that don't expect those;
getting pipeline back to preroll/playing again when audio/video streams
have different lengths and a seek past the end of one of the stream
happens doesn't always work, etc.). Needs further investigation in the
next cycle.
https://bugzilla.gnome.org/show_bug.cgi?id=633700https://bugzilla.gnome.org/show_bug.cgi?id=634699
Fix conversions to IYU1, they allocated infinite amounts of memory before
because no conversion to IYU1 was actually implemented and it was running
into an infinite loop trying to find suitable intermediate formats.
Also fix the stride and sizes used for IYU1.
Fix a bug when reconfiguring the playsink where the subpicture
stream is broken by attempting to connect it through
streamsynchroniser and second time.
Going over integer arithmetic will lead to minimal rounding errors,
leading to +/-1 changes for volume==1.0. Implement the controlled
processing with floating point arithmetic, which was already done
for the C versions anyway.
Advance stop times too when they are getting higher than the
stop time of segments, avoiding assertions.
The stop time has to be advanced too so that running time keep in sync
for gapless mode.
https://bugzilla.gnome.org/show_bug.cgi?id=631312
This moves AAC profile detection to pbutils, and uses this in
typefindfunctions. This will also be used in qtdemux.
https://bugzilla.gnome.org/show_bug.cgi?id=617314
API: gst_codec_utils_aac_get_profile()
API: codec_utils_aac_caps_set_level_and_profile()
This allows us to add generic codec-specific functionality, like
extracting profile/level data from headers, without having to duplicate
code across demuxers and typefindfunctions.
As a starting point, this moves over AAC level extraction code from
typefindfunctions, so it can be reused in qtdemux, etc.
https://bugzilla.gnome.org/show_bug.cgi?id=617314
API: gst_codec_utils_aac_get_sample_rate_from_index()
API: gst_codec_utils_aac_get_level()
Where it was previously located, we would get async-done for the first
unknown-type, even if other valid streams would appear afterwards.
decode_bin_expose() will take care of posting async-done when the group
is exposed.
But we still want to post it in case the typefinding returned an unknown
type, in which case we will post it after posting an error.
These two changes ensure we do as much as possible before posting async-done.
Replace moving-color-bars pattern with smpte100, and change
moving-speed to horizontal-speed. Default is now 0. Add
a rotation stage to pattern building.
Allocate a temporary scanline for building images. Remove
unused code. Disable several patterns that we're unable to
test and probably never used. Add other variants of bayer
sampling. Convert some patterns to use videotestsrc_blend_line.
Replace solid-color property with foreground-color and add
background-color. Pull some common code out of each of the
pattern generating functions. Fix many of the patterns to
use foreground-color/background-color instead of white/black.
Generated images are indentical to previously if foreground-color
and background-color are left as default.
API: GstVideoTestSrc::foreground-color
API: GstVideoTestSrc::background-color
Send FLUSH_STOP right after forwarding the seek event upstream if necessary.
This makes sure that adder->srcpad is not left flushing if seeking fails or if
upstream is blocked.
The same fix was already applied to videomixer in 49b2a946.
This should speed up standard Vorbis encoding and decoding pipelines a bit.
Thanks to David Schleef for the assistance to get the ORC code right
and explaining everything.
We currently don't use the GAP flag for video and the docs say
that this is for buffers, that have been created to fill a gap
and contains neutral data. For video this is the previous frame.
This information can be used by encoders to encode the duplicated
frames more efficiently. See bug #627459.
That is, if eos is received which will not be forwarded, and the stream
has not yet seen any data, then send a buffer to preroll downstream
(which might otherwise be accomplished by the eos event).
Streamsynchronizer excepts to see stream-changed msg for all streams, but to
arrange for this, video and subtitle streams need to be decoupled by means
of queues (due to pad blocks that may occur).
Fixes#626463.
Specifically, as the latter may have one thread pushing EOS to several streams,
that needs to be decoupled into various thread to prevent preroll hanging
problems.
Otherwise we're producing different caps and basetransform thinks that it
can't passthrough buffer allocations, etc.
In 0.11 all video caps really should have the PAR set...
... which generalizes the current listing of white, black, etc.
In particular, also allow specifying alpha channel, and modify
some structures and pattern filling to cater for alpha value as well.
Fixes#624919.
API: GstVideoTestSrc:solid-color
This fixes a race condition in playbin2's gapless mode, where the
EOS of other streams might arrive in the sinks before the last stream
ends and the switch to the new track happens. The EOS sinks won't
accept any new data then and playback stops.
To prevent this, delay all EOS events until all streams are EOS
and advance the sinks of the EOS streams by filler newsegment
events if necessary.
Fixes bug #625118.
This reads the 3gp profile from the major/compatible brands and puts
this as a 'profile' field in caps. This can be used by demuxers to
decide whether they can handle this stream or not. Also needed for
DLNA.
https://bugzilla.gnome.org/show_bug.cgi?id=620291
Logic for choice of GST_PAD_LINK_CHECK_* is as follows:
* Where return of pad_link wasn't checked before : NOTHING
* Where linking is between known compatible elements : NOTHING
* All other cases : TEMPLATE_CAPS
Slashes down playsink reconfigure by up to 50% cpu time.
This makes sure that we always keep the display aspect ratio and
add black borders if necessary, which is usually something you want
for viewing a video.
This behaviour was not preferred and caused visible image quality
degradations. The real solution would be, to apply a real
deinterlacing filter before scaling the frames.
Fixes bug #615471.
We only look for packets with payload, but it appears there may be packets without,
which makes it harder to find the N packets with payload in a row that we need in
order to typefind this successfully, so scan some more data than necessary in the
optimistic scenario. Alternatively we could change IS_MPEGTS_HEADER().
Fixes#623663.
Before gapless playback failed when switching between audio-only,
video-only and audio-video files, when choosing different clocks
and when the different streams had different durations.
This is now handled by a helper element, which keeps track of the
running times of all streams and synchronizes them.
Fixes bug #602437.
.weba (audio) and .webv (video) were speculation on my part before
the public launch. As of yet no decision has been made on the
file extension for audio-only WebM, and I'm pretty sure there will
never be one for video-only.
Fixes bug #623837.
Fixes spurious errors that happen after an error and playing a working
stream afterwards or signals that are emitted for non-active groups.
Fixes bug #624266.
This reverts commit 9d7538247f.
If the DVD subpicture caps are not part of the raw caps, uridecodebin
doesn't qualify resindvdbin as raw source and plugs decodebins, which
causes broken DVD playback because of bugs elsewhere.
This change was originally added to only expose supported, raw subtitles,
e.g. if the subtitle sink did not support DVD subpictures but a converter
to some supported format exists. It's not very important right now because
we have nothing (that is autoplugged) to convert from plaintext/pango-markup
or DVD subpictures to something else.
Fixes bug #623583.
Otherwise the uridecodebin will be still a child of playbin2 and
its signals will still be connected. In future state changes this
will then emit unrelated signals that will confuse playbin2 or,
even worse, cause crashes and assertions.
Fixes bug #623318.
If an error happens, the PAUSED state will never be reached. If an
application re-uses decodebin2 (like totem) where one would normally
set to READY between each file, the cleanup that normally happens in
the PAUSED=>READY codepath will never be called, resulting in the
following file to re-use the previous demuxer/decoder/...
https://bugzilla.gnome.org/show_bug.cgi?id=622807
We need to clear the pointer to our ts-offset element when we destroy the video
chain elements to make sure nobody derefs it to invalid memory afterwards.
Otherwise we would end up with a bogus ->audiochain->ts_offset field
which would cause segfaults/assertions when trying to modify the
'ts-offset' property in update_av_offset().
Was easy to trigger when using a list of audio+video files mixed with
video-only files in totem.
Use the pad caps when they are available to continue the autoplugging. If the
pad caps are set, they are fixed and then we can directly continue autoplugging.
Use an accumulator for the autoplug-sort signal so that we can stop the emission
when a signal handler produced a valid result. This avoids the object handler
to overwrite the results from user signals.
Fixes#621161
Scan a bit into the data when checking for dts frames instead
of expecting the frame sync to be right at the start of the
data. This is needed for some dts-disguised-as-pcm-in-wav files.
See #413942.
Orc is not a hard requirement. Things should still compile and
work without orc, but slow fallback code may be used in this
case. Fix up configure to not error out if orc is not installed
and wrap use of orc profiling in audioresample in #ifdefs.
Fixes#620136 some more.
Make jpeg typefinder check more than just the first two bytes
plus Exif or JFIF marker. This allows us to report MAXIMUM
probability in cases where there's no Exif or JFIF marker,
making typefinding stop early. Also extract width and height,
because we can.
Fix typo that made the AC-3 typefinder not actually check for a
second frame, but rather compare the sync point found to itself,
which resulted in the AC-3 typefinder reporting an overly optimistic
MAXIMUM or VERY_LIKELY probability when it found a possible frame
sync.
Move the convert_frame function to playsink and make it part of the API. This is
in preparation to add the convert_frame signal to playsink.
See #620279
If a file contains raw streams (not requiring a decoder) that we do
not want (expose-all-streams == FALSE), we would previously consider
those of unknown-type (missing a decoder) ... whereas in fact it was just
because they don't need decoders.
This only applies if expose-all-streams is FALSE.
* don't re-create our possible caps every single time, just use the
template caps.
* don't intersect the caps against the template, basetransform has already
done that for us.
62% speedup of _transform_caps() (instruction calls, measured with callgrind)
API : expose-all-streams
If disabled:
* only the streams that CAN be decoded and match the final caps will have a
decoder plugged in and be exposed.
* the streams that COULD HAVE BEEN decoded but do not match the finals caps
will not have a decoder plugged in and will not be exposed.
If no decoder is available to decode a certain stream, then the missing element
message will still be emitted regardless of the value of the property.
https://bugzilla.gnome.org/show_bug.cgi?id=617868
Adder was using always incrementing timestamps. Seeking was done by setting the
position in the newsegment event. This was failing when doing segmented seeks
with rate<0.0, as offset (and thus timestamp) would go below 0.
Now we take both cur and end from the seek event. We construct newsegment events
depending including cur and end from the seek event. We set position to the
start of the segment. Timestamp is set to start or end of segment depending on
rate. Offset is recalculated.
Use foo_LDADD instead of foo_LDFLAGS to specify the libraries to link to.
This should make sure arguments are passed to the linker in the right
order, and makes LDFLAGS usable again.
Based on initial patch by Brian Cameron <brian.cameron@oracle.com>
Fixes#615697.
This adds code to calculate the level for a given AAC stream and export
it in the stream caps. For AAC LC streams, the level is calculated
according to the definition under the AAC Profile. For other streams,
the definition under the Main Profile is used.
HE-AAC support is still to be done, and is dependent on detecting the
presence of SBR and PS in the stream.
Level is added as a field of type string because that's the way it's
done in H.264 caps as well. There are only a few possible levels, so
not using a numerical type is not too painful in this case, and
consistency is nice.
Fixes#613589.
This looks at the AAC profile for ADTS streams and adds the profile as a
string in the corresponding caps.
Profile is the actual profile, base-profile denotes the minimum codec
requirements to decode this stream. In this case they're always the
same, but they may differ e.g. in case of certain HE-AAC streams that
can be partially decoded by LC decoders (with loss of quality of course)
if no suitable HE-AAC decoder is available.
Fixes#612312.
Decrement sample counter when playing backwards. Set proper segment when playing
backwards (0..cur instead or cur..-1). Add more logging and fix a format string.
Unreffing it whenever the sinks are removed will make the volume
element unavailable after a playbin reuse because it is only
recreated if the audio sink has changed.
Fixes bug #614288.
In reverse mode we want use the next next timestamp (and not the other way
around). Fixes the tests again. Also readd a log line that was dropped with
previous commit.
We know our plugins and examples are independent of each other, so may
just as well build them in parallel. Makes the output a bit messy, but
that shouldn't be a problem and can easily be avoided with make -j1.
And fix the resulting compile failures.
I'm sorry about the patch necessary to gstclockoverlay.h but after
talking to Tim we decided we can live with it.
Change playbin2 to not error out if there are subtitles and audio
but no video. If visualizations are enabled the subtitles are rendered on top
of the visualization stream, otherwise the subtitles are not linked at all and
only the audio is played (and a warning message is posted).
If there are only subtitles but neither audio nor video an error message is
still posted.
Fixes bug #610866.
For this add subtitle encoding properties to playsink and subtitleoverlay
and update the values in the containing elements.
Also update the font description in textoverlay or the used renderer
element if it is changed during playback.
Fixes bug #610310.
Use the same translated message string for missing core elements as
playbin uses, which is a bit nicer and also indicates that there is
something wrong with the user's GStreamer installation (which arguably
is the case if elements like typefind or queue2 are missing).
Otherwise the ghostpad will still be linked to the peer and there
will still be a reference kept, leading to nothing being unlinked
and destroyed until decodebin2 is finalized.
This fixes reuse of decodebin2 if a raw stream is connected to
its sinkpad.
This makes sure that we don't destroy the last reference before the
element gets back to NULL state. Fixes assertion failures if a playbin2
instance is reused but different sinks are automatically chosen because
of different caps.
This reverts commit 7335ce5d3e.
Support abusing the uri property to configure the next uri to play
outside of the about-to-finish handler for the time being after all.
We also shouldn't use thread private structures for this, since it
should be possible to block the thread that emitted about-to-finish
while the main thread sets the uri property. See #607226.
When reusing a decodebin2 element, clear the properties we might have changed,
to their default values or else we might end up with old configuration.
Fixes#608484
Make AC-3 typefinder use the DataScanCtx stuff so we don't have to
do gst_type_find_peek() in the inner loop all the time. Also return
when we've suggested AC3 caps, instead of continuing with the loop.
When we are dealing with a source that produces raw audio/video, we don't use a
decodebin2 to decode the data and we thus don't have the drained/about-to-finish
signal emited. To fix this, we add a padprobe on the source pads and emit the
drained signal ourselves. This then makes playbin2 emit the about-to-finish
signal for raw sources such as cdda://
Fixes#607116
Add PNM typefinder, so we can remove the one that's in the PNM plugin
in -bad (which btw uses different/wrong media types that don't match
the ones used by gdkpixbufdec) and people don't make fun of us for
loading image decoders when typefinding and playing back audio files.
We don't want to end up setting values on elements where the property is of
a different type than we expect. Can't transform the value either, since we
can't really make assumptions about the scale and transform function.
Fixes crashes when using playbin2 with apexsink (#606949).
Changing the URIs in a state > READY results in unexpected behaviour,
i.e. the new URIs are only used after the current track has finished.
Fixes bug #607226.
In this case the video still goes through the text chain and
subtitles are still going in there, in case subtitles are
enabled again. This makes sure that re-enabling subtitles
happens instantly.
Fixes hanging video when disabling subtitles, caused by an
unliked video pad.
Detect EOS faster.
Try to reuse one of the input buffer as the output buffer. This usually works
and avoids an allocation and a memcpy.
Be smarter with GAP buffers so that they don't get mixed or cleared at all. Also
try to use a GAP buffer as the output buffer when all input buffers are GAP
buffers.
It may not be uncommon for the input timestamps to experience some jitter
around the 'perfect time'. As such, instead of regularly adding and dropping
samples, optionally allow for some tolerance in a more relaxed approach.
API: GstAudioRate:tolerance
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.
