We only check input from the API user with g_return_*_if_fail().
Internal sanity checks should use g_assert() instead, which is
disabled by default for releases.
* Avoid repeating code everywhere, and instead provide all parsing
information in one go.
* Add BAT support
* Refine BAT/CAT identification (by adding PID checks)
Fix calculation of the frame cropping rectangle, and more precisely
the actual cropped height. The frame_crop_top_offset subtraction
was not scaled up with SubHeightC.
Also clean-up variables to align more with (7-18) to (7-21).
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
Assign the un-cropped width/height to sps->width/sps->height
during sps header parsing. Added new fields to SPS header structure
to provide the crop-rectangle dimensions.
https://bugzilla.gnome.org/show_bug.cgi?id=694068
Cancelled is a 'permanent' state of the uridownloader and is only
removed by a call to _reset. When a download fails we just want to
return NULL on the fetch function and leave the downloader ready
for another fetch, otherwise the user has to call _reset after
failed downloader, even when it didn't call _cancel.
Due to the variety of section types out there, we need to add
some checks when identifying section types.
We check here that the PID is also consistent with the table_id.
The size checks were wrong. The smallest size for a NIT is 16 bytes
(12 for the smallest content + 4 for crc) and the smallest size for
a inner stream loop is 6 bytes (without any descriptors).
Also remove FIXME that has already moved elsewhere
Add API to parse the Slice header. This also calculates the macroblock
position as specified in 6.3.16.
https://bugzilla.gnome.org/show_bug.cgi?id=664274
Signed-off-by: Sreerenj Balachandran <sreerenj.balachandran@intel.com>
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
Add new interface to MPEG-2 video parser that takes GstMpegVideoPacket
arguments instead of data, size, and offset. New functions are called
after gst_mpeg_video_packet_*() and provide the default implementation.
Older API is moved to the deprecated namespace and uses the new functions.
https://bugzilla.gnome.org/show_bug.cgi?id=692933
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
* Add a base page for the library
* Add pages for the base MPEG-TS section and descriptors
* Add pages for the known variants
* Add documentation on more fields/sections/types
* Remove some fixmes that were ... fixed
These are the values officially registered in the base specification
(H.222.0/13818-1). Later on we can add other enums for other variants
Note that the enum is not used in the structure fields (such as a pmt
stream stream_type field) since it can contain values from other
variants.
* In order to avoid future clashing between table_id for the various
mpeg-ts variants, use different enums.
* In order to keep everything clean(ish) and allow for cleaner growth,
split into different files (will need the same for descriptors later)
* Also ... implement free functions for all table types :)
Sorry for this :( But this makes it more in sync with expected type
naming in gobject (i.e. CamelCase and not CamelMAYBECase).
Also split descriptor type enums into the different variants:
* ISO H.222.0 / 13818-1 (i.e. standard mpeg-ts)
* DVB
* ATSC
* ISDB
* miscellaneous
This will avoid future clashes when specs use the same descriptor type
Exposes various MPEG-TS (ISO/IEC 13818-1) and DVB (EN 300 468) Section
Information as well as descriptors for usage by plugins and applications.
This replaces entirely the old GstStructure-based system for conveying
mpeg-ts information to applications and other plugins.
Parsing and validation is done on a "when-needed" basis. This ensures
the minimal overhead for elements and applications creating and using
sections and descriptors.
Since all information is made available, this also allows applications
to parse custom sections and descriptors.
Right now the library is targeted towards parsing, but the structures
could be used in the future to allow applications to create and inject
sections and descriptors (for usage by various mpeg-ts elements).
https://bugzilla.gnome.org/show_bug.cgi?id=702724
Adds a new API gst_uri_downloader_fetch_uri_with_range that allows
downloading only a byte range from an URI. It uses a seek event
sent to the source to signal the range to be downloaded.
https://bugzilla.gnome.org/show_bug.cgi?id=702206
Ignore the display_extension values if they are greater than the width/height
values provided by seqhdr and calculate the PAR based on the seqhdr values.T
his is what DVD players are doing.
Thanks to "David Schleef <ds@schleef.org>"
https://bugzilla.gnome.org/show_bug.cgi?id=685103
In some scenarios, for example in QtWebKit, might be difficult to obtain full
control on the egl display and it might be only accessible indirectly via
eglGetCurrentDisplay().
https://bugzilla.gnome.org/show_bug.cgi?id=700058
When chain method was called after gst_uri_downloader_stop and before state has been changed to NULL, execution was blocking on g_mutex_lock.
Conflicts:
gst-libs/gst/uridownloader/gsturidownloader.c
When downloading and cancelling quickly the uridownloader object and the
element using it could miss the cancelled window and the uridownloader
would fetch the wrong URI and block on subsequent fetches.
This was also problematic when stopping elements, while one task would
call the cancel, another element thread could issue a new fetch_uri. As
the cancel state isn't 'permanent' this fetch_uri would block and
prevent the whole element from stopping and going to NULL.
This patch makes the 'cancelled' state permanent until a
gst_uri_downloader_reset is called. This way the element knows the
window where the uridownloader isn't active and only reactivate it when
ready.
This can be used by parsers to provide pre-parsed information to
downstream elements that would require it (so they can avoid having
to parse the bitstream again).
The content of the EGLImages can be at least in GStreamer orientation,
meaning top line first in memory, or OpenGL orientation, meaning
bottom line first in memory.
Add utility functions to convert quantization matrices from zigzag scan
order (as encoded in the bitstream) into raster scan order. Also provide
another function to reverse the operation.
https://bugzilla.gnome.org/show_bug.cgi?id=693000
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
Quantizer matrices are encoded in zigzag scan order in the bitstream,
but they are stored in raster scan order when they are parsed. However,
default matrices were also prepared in zigzag scan order, hence the
mismatch. i.e. the matrices were presented either in raster scan order
if they are explicitly present in the bitstream, or they were presented
in zigzag scan order if the default definitions were to be used instead.
One way to solve this problem is to always expose the quantization
matrices in zigzag scan order, since this is the role of the parser to
not build up stories from the source bitstream and just present what
is in there.
Utility functions will be provided to convert quantization matrices in
either scan order.
https://bugzilla.gnome.org/show_bug.cgi?id=693000
Signed-off-by: Cong Zhong <congx.zhong@intel.com>
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
Fix parsing of residual bytes. This is a two-step process. First,
remaining colums of full vertical resolution (<height>) need to be
processed. Next, remaining bytes in the first row can be processed,
while taking into account the fact that we may have filled in the
first columns already. So, this is not full horizontal resolution.
The following figure helps in understanding the expected order of
operations, for a 8x5 MBs bitplane.
5 5 6 6 6 6 6 6
5 5 1 1 1 2 2 2
5 5 1 1 1 2 2 2
5 5 3 3 3 4 4 4
5 5 3 3 3 4 4 4
So, after tiles 1 to 4 are decoded, vertical tile 5 needs to be
processed (2x5 MBs) and then the horizontal tile 6 (6x1 MBs).
https://bugzilla.gnome.org/show_bug.cgi?id=692461
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
Fix decoding of DIFF6 or NORM6 bitplanes with an odd number of lines
(3x2 "horizontal" tiles). In this case, we have to skip the first line
of macroblocks but <width> number of bytes was used to do so, instead
of the actual <stride> size.
This fixes decoding for the video sample attached to:
https://bugzilla.gnome.org/show_bug.cgi?id=668565https://bugzilla.gnome.org/show_bug.cgi?id=692461
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
Add gst_vc1_parse_slice_header() function to parse slice headers as
described in 7.1.2. Slice layers are optional and allowed in advanced
profile mode only. Picture header, if available (PIC_HEADER_FLAG),
is parsed but not recorded because it shall be the same as that was
previously parsed with gst_vc1_parse_frame_header().
This fixes SA00049.vc1 conformance test.
https://bugzilla.gnome.org/show_bug.cgi?id=692388
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
Fix decoding of DIFF2 or NORM2 bitplanes with an odd number of macroblocks.
In particular, account for the first bit that was already parsed so that to
avoid a buffer overflow after all pairs are parsed.
This fixes SA00040.vc1 conformance test.
https://bugzilla.gnome.org/show_bug.cgi?id=692312
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
Rename dqsbedge to dqbedge. The intent is that we can only have a single
boundary edge selector, depending on the value of dqprofile. So, dqbedge
represents DQSBEDGE if dqprofile == GST_VC1_DQPROFILE_SINGLE_EDGE, or
DQDBEDGE if dqprofile == GST_VC1_DQPROFILE_DOUBLE_EDGE.
The former dqbedge field is marked as unused and can be removed on the
next gst-plugins-bad version that allows ABI changes.
https://bugzilla.gnome.org/show_bug.cgi?id=692272
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
Fix parsing of VOPDQUANT when DQUANT == 2. In particular, DQUANTFRM is
not present in the bitstream in this case and it shall be derived to
the default value of zero (7.1.1.31.1).
https://bugzilla.gnome.org/show_bug.cgi?id=692271
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
Fix calculation of ALTPQUANT when DQUANT == 1. PQDIFF alters ALTPQUANT
in any case. See 7.1.1.31.6.
https://bugzilla.gnome.org/show_bug.cgi?id=692270
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
Fix parse_vopdquant() to correctly parse DQPROFILE, which is 2 bits
instead of a single bit.
https://bugzilla.gnome.org/show_bug.cgi?id=692267
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
The standard specifies that when slice_beta_offset_div2 is not present
in the slice header, then the value of slice_beta_offset_div2 shall be
inferred to be equal to 0.
https://bugzilla.gnome.org/show_bug.cgi?id=692265
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
Coverity found missing break in parse_frame_header_advanced() when
determining PTYPE from FPTYPE for interlaced streams.
https://bugzilla.gnome.org/show_bug.cgi?id=688626
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
gst_h264_parse_sps() returned FALSE if it parsed invalid (negative)
size components. Now make it gracefully return GST_H264_PARSER_ERROR
instead of GST_H264_PARSER_OK (FALSE).
https://bugzilla.gnome.org/show_bug.cgi?id=684568
Change the way the pixel-aspect-ratio is computed by
interpreting the sequence header aspect ratio info
as MPEG-1 values until a sequence extension or
sequence display extension is seen, and then updating
the sequence header struct accordingly.
Fixes incorrect anamorphic display on some MPEG-2 (DVD)
sequences.
ASPECT_HORIZ_SIZE and ASPECT_VERT_SIZE are syntax elements that hold
binary encodings of sizes ranging from 1 to 256. Thus, the calculated
pixel-aspect-ratio was off by one.
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
https://bugzilla.gnome.org/show_bug.cgi?id=683858
Anonymous union is an ISO C (2011) feature that is not exposed in
compilers strictly conforming to the previous standard.
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
need to pass a GstSample to the utilitary preview buffer post functions
as a GstBuffer doesn't have caps anymore. The GstSample has the GstCaps
and it is used to inform the preview's pipeline about the format of the
input, before it gets converted to the user's requested output format.
... to allow for more efficient parsing and (more) consistent parsing API
among various codec parsers.
Fixes#672701.
Conflicts:
gst/videoparsers/gstmpegvideoparse.c
This always happens with GstByteReader/Writer and friends when
not taking into account returned boolean of the _read/_write functions
(which is actually wrong).
Make use of the *_unchecked variant as much as possible, or take the
returned value into account.
Some hardware video decode acceleration API (VA-API, DXVA) require
a bit count to the first macroblock, minus the number of emulation
prevention bytes. So, instead of having the consumer of the library
scan the slice_header() again, just record that number while parsing.
Signed-off-by: Gwenole Beauchesne <gwenole.beauchesne@intel.com>
https://bugzilla.gnome.org/show_bug.cgi?id=671203
The entries were not filled in linearly and the termination was not
recorded either. Now, the actual number of modifications is recorded
similarly to dec_ref_pic_marking(). i.e. an explicit counter instead
of storing the termination value in the array.
https://bugzilla.gnome.org/show_bug.cgi?id=668192
altref/invisible 'frame' do not describe some frame directly, so it can't be
displayed and timestamps should not be updated.
Fix bug: https://bugzilla.gnome.org/show_bug.cgi?id=655245
Signed-off-by: Oleksij Rempel (Alexey Fisher) <bug-track@fisher-privat.net>
New caps will need to be negotiated when reset happens
(PAUSED to READY). Without reseting the internal
stored format, basevideoencoder/decoder wouldn't call the
configuration function when the same caps was negotiated
again as they would believe this was the same caps as before.
The issue is that _stop has been called when going to READY and
the elements would have reset their internal codec libs/state as
well. A new configuration should be done.
This is useful for cases where the caller *knows* that the provided
input contains a whole NALU and can therefore avoid:
* the expensive checks for the next start code (which won't be present)
* delaying the input parsing (since we would need the next incoming NALU
in order for the parsing code to detect the next start code)
https://bugzilla.gnome.org/show_bug.cgi?id=665584
Const-ify one more VLC table. Fix spelling of 'hybrid'.
No need to explicitly call ensure_debug_category() everywhere,
that will be done automatically from GST_LOG() and friends
via GST_CAT_DEFAULT.
Get rid of weird code that copies a list manually, taking
ownership of the elements and then frees the old list. Instead,
just take over the old list entirely. (If the intent was to
reverse the list, one could use g_list_reverse() instead).
Then, push events in the list out from last to first (since they
were prepended as they came in) instead of just pushing out the
last in the list and leaking the others.
When vui->timing_info_present is 0, vui->fixed_frame_rate_flag and others
cannot be accessed since they have not been set.
It was also possible that sps->fps_{num,den} end up initialized here.
Create preview pipeline already in initialization phase. This speeds
up NULL_TO_READY state change. Also implement a separate function for
setting the preview filter element.
This also restricts the preview filter property to work only on
NULL state.
Instead of having a single VC1SequenceHeader structure, use the 3 structs
from the "Table 265: Sequence Layer Data Structure" of the specification
for the library to be more flexible.
Implement the functions to parse them
Use g_new0 to initialize all fields with 0 to only cleanup what has been
initialized. This makes cleanup work correctly when some initialization
fails and pointers are left in some inconsistent state.
If the preview pipeline fails creation, for any reason, we should
fail basecamerasrc state change.
Also adds a missing g_return_if_fail check to preview pipeline
functions
The preview pipeline doesn't need 2 colorspace converters, remove
one to speed up caps negotiation and reduce the delay on getting
the first preview buffer out of the preview pipeline.
It shouldn't cause problems as videoscale and ffmpegcolorspace seems
to handle the same caps, so no conversion should be needed for
videoscale. Additionally, camerabin1 has been working with a similar
pipeline with a single ffmpegcolorspace and no bugs have been open about it
so far.
This allows to specify constraints on the compressed downstream caps
by muxers or capsfilters, which will then be forwarded to upstream
and allows video converters to fulfill the constraints.
Code based on Mark Nauwelaerts audio encoder base class.
GstPhotography API contains functions to get/set flicker reduction
mode, but GstPhotoCaps enumeration doesn't have item for it, so elements
are not able to report whether they support this feature or not.
Also add useful GST_PHOTOGRAPHY_CAPS_ALL for easily selecting all
capabilities at once.
https://bugzilla.gnome.org/show_bug.cgi?id=655318
The use of this method was removed in:
commit 539f10f4d9
basecamerasrc: More cleanup
The code from wrappercamerabinsrc is from v4l2camerasrc but is unused:
get_allowed_input_caps is not called anywhere.
Implements a message handling function to preview pipeline bus.
If GST_MESSAGE_ERROR is seen, considers preview pipeline unable
to do its job and posts an error message to application.
Sets pipeline element to NULL so that subsequent calls to post_preview
and set_caps functions just returns without pushing anything to the
disposed preview pipeline. Leaves further actions to the application.
Implements a state indicating flag to preview pipeline,
so that new caps are not set if the pipeline is processing a
preview. The caps are set as pending and applied when the
next preview post is called.
In this case a wait was implemented in the post_preview function,
so that new preview image buffer will wait until the other previews
have been posted to the application and the new caps can be used
safely.
While this changes API slightly (e.g. actually uses set_format now), which is OK
for unstable API, it has following merits:
* symmetric w.r.t. stop at state change
* in line with other base class practice
* otherwise no subclass method at state change (global activation time)
Moreover, subclassese are either unaffected or trivially adjusted accordingly.
While this changes order w.r.t. set_format, which is OK for unstable API,
it has following merits:
* symmetric w.r.t. stop at state change
* in line with other base class practice
* little benefit in invoking 2 subclass virtual methods (set_format and start)
in immediate succession; all actions in the second could be done in the first
whereas subclass has no chance to do anything 'global' at activation time
Moreover, current -bad subclass relevant methods either trivially commute
or are either trivially adjusted accordingly.
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
This is not implemented in any of our real sources to which wrappercamerabinsrc
might connect but this is optional and can be implemented at any time. A
limit on the software zoom level using video{crop,scale} would be arbitrary.
Use resource warning messages to notify camerabin2 that a capture
as aborted or couldn't be started, making it decrement the
processing counter and making the idle property more reliable.
Checks if the new received preview-caps are equal to what is
already in use, skips the preview-caps setting logic in case
new caps are same as current ones.
Adds a virtual function to basecamerasrc in case subclasses want to be
notified of changing preview caps. This is useful if the subclass wants
to post the preview itself or if it wants to provide a preview buffer
as close to as possible to the user's requested resolution to the
preview generation pipeline.
Adds some more logging and always assume capture has started before
start_capture is called. This helps on image captures that might
call finish_capture directly from start_capture or before start_capture
finishes.
Adds a readable property to gstphotography interface to query
what are the allowed preview caps supported.
Patch by Tommi Myöhänen <ext-tommi.1.myohanen@nokia.com>
The timestamps are only used if the output adapter is used, not
if complete frames are provided by the decoder and finish_frame() is
called and even in the case where the output adapter is used they
might not be used and are leaked.
Add some guards and fat warnings to the header files with still unstable
API, so people who just look at the installed headers know that it
actually is unstable API.
Merging previous commit into current codebase.
Move include directives for gst-libs into GST_PLUGINS_BAD_CFLAGS,
and fix all the Makefiles that use it. This is so that all the
include directories are added in the proper order: first the
directories in srcdir/builddir, then gst-plugins-base dirs, then
gstreamer dirs. If the order is wrong, installed headers may be
used instead of local headers and/or uninstalled headers from -base.
Because config.h defines __MSVCRT_VERSION__, which should be defined
before inclusion of any system header.
Also fixes mpegdemux Makefile.am LIBADD typo.
Fixes#606665
This allows to get rid of the sampling_rate variable in the base-class. Also now
subclasses can modify the caps to actualy negotiate. This is needed to e.g. set
audio-channel positions.
Revert the changes that added audio positions to template caps. We have an un-
fortunate limitation in core that does not allow to do it. Keep a few things
commented out, so that the channel position can later on be set in setcaps.
This reverts commit 4c087bcb07.
The reverted commit changes the order that set_format() and start()
are called, which is incorrect. The correct order is set_format(),
start(), handle_frame()..., stop()
This queries port roles from the LV2 data and converts it into GStreamer
channel positions. This should allow any type of multi-channel plugin
(including beyond stereo, e.g. surround) to work fine in GStreamer,
and with elements that require channel positions to be explicitly stated.
Install the headers, version the library with @GST_MAJORMINOR@,
add all required libraries to _LIBADD instead of _LDFLAGS,
and add GST_*_LDFLAGS to _LDFLAGS.
Fixes bug #594715.
-remove gst-libs/gst/dshow
-fakesource is moved from gst-libs/gst/dshow to sys/dshowsrcwrapper
-some minor changes (C/C++ check and includes) to make the plugin
compile again.
Add some guards and fat warnings to the header files with still unstable
API, so people who just look at the installed headers know that it
actually is unstable API.
Also move schroedinger plugin. This creates a new library,
gstbasevideo-0.10, which will probably be merged back into
gstvideo-0.10 when this is moved back to -base.
- Separate gstsignalprocessor into a separate library (not sure if this
is in the right place, but it works for now anyway)
- Create LV2 element based on LADSPA element, port most discovery
functionality
Application developers won't know right away which module an interface comes from,
and may assume that it is covered by the usual GStreamer API guarantees, so make
it as clear as possible that this particular API is still subject to change
(should have done that with other libraries in -bad before too really).
Original commit message from CVS:
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_init), (gst_app_src_set_property),
(gst_app_src_get_property), (gst_app_src_query),
(gst_app_src_set_latencies), (gst_app_src_set_latency),
(gst_app_src_get_latency), (gst_app_src_push_buffer_full):
* gst-libs/gst/app/gstappsrc.h:
Add properties and methods to configure and retrieve the min and max
latencies.
Original commit message from CVS:
* examples/app/appsrc-ra.c: (feed_data):
* examples/app/appsrc-seekable.c: (feed_data):
* examples/app/appsrc-stream.c: (read_data):
* examples/app/appsrc-stream2.c: (feed_data):
Fix example to unref after emiting the push-buffer action.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_push_buffer_full), (gst_app_src_push_buffer),
(gst_app_src_push_buffer_action):
Don't take the ref on the buffer in push-buffer action because it's too
awkward for bindings. Fixes#564482.
Original commit message from CVS:
Patch by: 이문형 <iwings at gmail dot com>
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_push_buffer):
Don't forget to release the lock again if we bail out because some
pad is flushing or we've reached EOS, otherwise things will lock up
next time _push_buffer() is called (#562802).
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsrc.c:
* gst/h264parse/gsth264parse.c:
Wim, you're a bad boy. You don't want people to contact you or what?
Original commit message from CVS:
2008-06-16 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes)
(gst_app_src_get_max_bytes, gst_app_src_push_buffer): Use
G_GUINT64_FORMAT. Avoid overflow in get_max_bytes().
Original commit message from CVS:
* examples/app/.cvsignore:
* examples/app/Makefile.am:
* examples/app/appsink-src.c: (on_new_buffer_from_source),
(on_source_message), (on_sink_message), (main):
Add beefed up example app from bug #413418. It now also uses appsink
instead of fakesink for more ultimate coolness.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_init), (gst_app_src_set_property),
(gst_app_src_get_property), (gst_app_src_unlock),
(gst_app_src_unlock_stop), (gst_app_src_create),
(gst_app_src_set_max_bytes), (gst_app_src_push_buffer),
(gst_app_src_end_of_stream):
* gst-libs/gst/app/gstappsrc.h:
Add block property to allow push based implementation to block when we
fill up the appsrc queues.
Emit the enough-data signal while releasing our lock.
Original commit message from CVS:
* ext/dc1394/gstdc1394.c:
* ext/ivorbis/vorbisdec.c:
* ext/jack/gstjackaudiosink.c:
* ext/metadata/gstmetadatademux.c:
* ext/mythtv/gstmythtvsrc.c:
* ext/theora/theoradec.c:
* gst-libs/gst/app/gstappsink.c:
* gst/bayer/gstbayer2rgb.c:
* gst/deinterlace/gstdeinterlace.c:
* gst/rawparse/gstaudioparse.c:
* gst/rawparse/gstvideoparse.c:
* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/gstrtpssrcdemux.c:
* gst/selector/gstinputselector.c:
* gst/selector/gstoutputselector.c:
* gst/videosignal/gstvideoanalyse.c:
* gst/videosignal/gstvideodetect.c:
* gst/videosignal/gstvideomark.c:
* sys/oss4/oss4-mixer.c:
* sys/oss4/oss4-sink.c:
* sys/oss4/oss4-source.c:
Do not use short_description in section docs for elements. We extract
them from element details and there will be warnings if they differ.
Also fixing up the ChangeLog order.
Original commit message from CVS:
* examples/app/Makefile.am:
* examples/app/appsrc-ra.c: (feed_data), (seek_data),
(found_source), (bus_message), (main):
* examples/app/appsrc-seekable.c: (feed_data), (seek_data),
(found_source), (bus_message), (main):
* examples/app/appsrc-stream2.c: (feed_data), (found_source),
(bus_message), (main):
Added 3 more example application for using appsrc in random-access mode,
pull-mode streaming and pull mode seekable.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_start), (gst_app_src_do_get_size),
(gst_app_src_create):
* gst-libs/gst/app/gstappsrc.h:
Make stream-type property writable.
Unset flushing when starting so that we reuse appsrc.
Inform basesrc about the configured size.
Emit seek-data signal when we are going to a different offset in
random-access mode.
Original commit message from CVS:
* examples/app/.cvsignore:
* examples/app/Makefile.am:
* examples/app/appsrc-stream.c: (read_data), (start_feed),
(stop_feed), (found_source), (bus_message), (main):
Added an example on how to use appsrc in playbin in streaming mode from
an mmapped file.
* examples/app/appsrc_ex.c: (main):
Set pipeline to NULL to free queued buffers.
* gst-libs/gst/app/gstapp-marshal.list:
* gst-libs/gst/app/gstappsrc.c: (stream_type_get_type), (_do_init),
(gst_app_src_class_init), (gst_app_src_init),
(gst_app_src_flush_queued), (gst_app_src_dispose),
(gst_app_src_set_property), (gst_app_src_get_property),
(gst_app_src_unlock), (gst_app_src_unlock_stop),
(gst_app_src_start), (gst_app_src_stop), (gst_app_src_is_seekable),
(gst_app_src_check_get_range), (gst_app_src_do_seek),
(gst_app_src_create), (gst_app_src_set_stream_type),
(gst_app_src_get_stream_type), (gst_app_src_set_max_bytes),
(gst_app_src_get_max_bytes), (gst_app_src_push_buffer),
(gst_app_src_end_of_stream), (gst_app_src_uri_get_type),
(gst_app_src_uri_get_protocols), (gst_app_src_uri_get_uri),
(gst_app_src_uri_set_uri), (gst_app_src_uri_handler_init):
* gst-libs/gst/app/gstappsrc.h:
Measure max queue size in bytes instead.
Add support for 3 modes of operation, streaming, seekable and
random-access, making basesrc handle the scheduling modes for each.
Add appsrc:// uri handler so that automatic plugging can be done from
playbin2 or uridecodebin, for example.
Added support for custom segment formats.
Add support for push and pull based operations from the application.
Expand the methods so that errors can be detected.
Flush the queued buffers on seeks and when shutting down.
Add signals to inform the app that a seek must happen.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
(gst_app_sink_init), (gst_app_sink_set_property),
(gst_app_sink_get_property), (gst_app_sink_unlock_start),
(gst_app_sink_unlock_stop), (gst_app_sink_flush_unlocked),
(gst_app_sink_start), (gst_app_sink_stop), (gst_app_sink_event),
(gst_app_sink_preroll), (gst_app_sink_render),
(gst_app_sink_set_caps), (gst_app_sink_set_drop),
(gst_app_sink_get_drop):
* gst-libs/gst/app/gstappsink.h:
Start some docs.
Add property to drop buffers when the queue is filled
Fix unlocking and flushing when the queues are filled.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
(gst_app_sink_init), (gst_app_sink_set_property),
(gst_app_sink_get_property), (gst_app_sink_event),
(gst_app_sink_preroll), (gst_app_sink_render),
(gst_app_sink_set_emit_signals), (gst_app_sink_get_emit_signals),
(gst_app_sink_set_max_buffers), (gst_app_sink_get_max_buffers),
(gst_app_sink_pull_buffer):
* gst-libs/gst/app/gstappsink.h:
Add more docs.
Add signals for when preroll and render buffers are available.
Add property to control signal emission.
Add property to control the max queue size.
Original commit message from CVS:
* gst-libs/gst/dshow/Makefile.am:
Use CXXFLAGS rather than CFLAGS; these are C++ files.
Define required constants appropriately.
* sys/dshowdecwrapper/Makefile.am:
Add required include dir, libraries.
Define required constants appropriately.
Original commit message from CVS:
* gst-libs/gst/dshow/Makefile.am:
* sys/dshowdecwrapper/Makefile.am:
* sys/dshowsrcwrapper/Makefile.am:
Add Makefiles to win32 plugins and lib.
They will need to be tested and probably fixed by developers
working with mingw. This is a first step to include source files
with releases.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose):
Really clean up the queue instead of just unreffing all buffers
in it.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_base_init),
(gst_app_src_class_init), (gst_app_src_init),
(gst_app_src_dispose), (gst_app_src_finalize):
Fix dispose/finalize.
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* gst-libs/gst/dshow/gstdshowfakesink.cpp:
(CDshowFakeSink.CDshowFakeSink):
* gst-libs/gst/dshow/gstdshowfakesink.h: (CDshowFakeSink.m_hres):
Fix crasher in constructor due to the base class's constructor
not necessarily being NULL-safe (depends on the SDK version used
apparently; #492406).
* sys/dshowsrcwrapper/gstdshowaudiosrc.c: (gst_dshowaudiosrc_prepare):
* sys/dshowsrcwrapper/gstdshowvideosrc.c: (gst_dshowvideosrc_set_caps):
Fix a couple of MSVC compiler warnings (#492406).
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_marshal_OBJECT__VOID),
(gst_app_sink_class_init), (gst_app_sink_init),
(gst_app_sink_dispose), (gst_app_sink_finalize),
(gst_app_sink_set_property), (gst_app_sink_get_property),
(gst_app_sink_flush_unlocked), (gst_app_sink_start),
(gst_app_sink_event), (gst_app_sink_getcaps),
(gst_app_sink_set_caps), (gst_app_sink_get_caps),
(gst_app_sink_is_eos), (gst_app_sink_pull_preroll),
(gst_app_sink_pull_buffer):
* gst-libs/gst/app/gstappsink.h:
Add properties, signals and actions to access the element even without
linking to the library.
Fix some method names and signatures.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init):
Override the preroll vmethod instead of overriding the render method
twice.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_base_init),
(gst_app_sink_class_init), (gst_app_sink_dispose),
(gst_app_sink_flush_unlocked), (gst_app_sink_start),
(gst_app_sink_stop), (gst_app_sink_event), (gst_app_sink_preroll),
(gst_app_sink_render), (gst_app_sink_get_caps),
(gst_app_sink_set_caps), (gst_app_sink_end_of_stream),
(gst_app_sink_pull_preroll), (gst_app_sink_pull_buffer):
* gst-libs/gst/app/gstappsink.h:
Make love to appsink.
Make it support pulling of the preroll buffer.
Add docs and debug statements.
Fix some races wrt to EOS handling and stopping.
Implement getcaps.
Implement FLUSHING.
API: gst_app_sink_pull_preroll()
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins.args:
Remove directsoundsink property doc as this sink use the mixer
interface now.
* docs/plugins/gst-plugins-bad-plugins.interfaces:
Add interfaces implemented by Windows sinks.
* sys/directsound/gstdirectsoundsink.c:
* sys/directsound/gstdirectsoundsink.h:
Remove directsoundsink property and implement the mixer interface.
* win32/vs6/gst_plugins_bad.dsw:
* win32/vs6/libgstdirectsound.dsp:
Update project files.
* gst-libs/gst/dshow/gstdshow.cpp:
* gst-libs/gst/dshow/gstdshow.h:
* gst-libs/gst/dshow/gstdshowfakesink.cpp:
* gst-libs/gst/dshow/gstdshowfakesink.h:
* gst-libs/gst/dshow/gstdshowfakesrc.cpp:
* gst-libs/gst/dshow/gstdshowfakesrc.h:
* gst-libs/gst/dshow/gstdshowinterface.cpp:
* gst-libs/gst/dshow/gstdshowinterface.h:
* win32/common/libgstdshow.def:
* win32/vs6/libgstdshow.dsp:
Add a new gst library which allow to create internal Direct Show
graph (pipelines) to wrap Windows sources, decoders or encoders.
It includes a DirectShow fake source and sink and utility functions.
* sys/dshowsrcwrapper/gstdshowaudiosrc.c:
* sys/dshowsrcwrapper/gstdshowaudiosrc.h:
* sys/dshowsrcwrapper/gstdshowsrcwrapper.c:
* sys/dshowsrcwrapper/gstdshowsrcwrapper.h:
* sys/dshowsrcwrapper/gstdshowvideosrc.c:
* sys/dshowsrcwrapper/gstdshowvideosrc.h:
* win32/vs6/libdshowsrcwrapper.dsp:
Add a new plugin to wrap DirectShow sources on Windows.
It gets data from any webcam, dv cam, micro. We could add
tv tunner card later.
Original commit message from CVS:
* gst-libs/gst/app/Makefile.am:
Use GST_ALL_LDFLAGS, which actually exists, but maybe David
can confirm that was what he wanted.
Original commit message from CVS:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/app/gstappbuffer.c:
* gst-libs/gst/app/gstappbuffer.h:
* gst-libs/gst/app/gstappsrc.c:
Add GstAppBuffer that includes a callback and closure for
proper handling of data chunks.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_audio_caps_with_data):
Do actually fix invalid RIFF fmt header values for alaw
and mulaw audio instead of just saying so.
* gst/wavparse/gstwavparse.c: (gst_wavparse_fmt):
Give gst_riff_create_audio_caps_with_data() a chance to
fix up broken format header fields before extracting any
parameters from the header. (fixes#167633)
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data):
Add extradata to huffyuv (fixes#165013).
* gst-libs/gst/riff/riff-read.c:
(gst_riff_read_strf_vids_with_data):
Fix extradata extraction if it is in the chunk size.
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (gst_riff_peek_element_data),
(gst_riff_read_element_data):
* gst-libs/gst/riff/riff-read.h:
Add _peek version (req'ed in CDXA).
* gst/cdxaparse/gstcdxaparse.c: (gst_cdxaparse_init),
(gst_cdxaparse_loop):
Fix parsing in playbin.
* gst/playback/gstdecodebin.c: (close_pad_link):
Ignore current_ pads, they cause major annoyance.
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_use_event):
Don't bail on unknown events.
* gst/audioscale/gstaudioscale.c: (gst_audioscale_chain):
Don't crash on events before negotiation.
* gst/avi/gstavidemux.c: (gst_avi_demux_add_stream):
Send tags on pads, too.
* gst/playback/gststreamselector.c:
(gst_stream_selector_request_new_pad):
Forward events on first pad if no input was selected yet.
Original commit message from CVS:
* gst-libs/gst/riff/riff-ids.h:
* gst/wavenc/riff.h:
Add AMR (VBR and CBR) ids to riff.h audio codec list
* gst/asfdemux/gstasfdemux.c:
(gst_asf_demux_process_ext_content_desc),
(gst_asf_demux_process_object):
Retrieve more tags from ASF files (Genre, AlbumTitle, Artist)
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* gst-libs/gst/resample/resample.c: (gst_resample_sinc_ft_s16):
Fix invalid memory access (#159211).
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data):
Add BLZ0 (Blizzard's version of DivX) fourcc.
Original commit message from CVS:
* configure.ac: add audioresample and cairo plugins. Remove
HAVE_MMX stuff, because it's not used.
* ext/Makefile.am: same
* ext/audioresample/Makefile.am: You are not ready for an
audio resampling element based on audioresample.
* ext/audioresample/gstaudioresample.c:
* ext/audioresample/gstaudioresample.h:
* ext/cairo/Makefile.am: You are not ready for overlay elements
based on cairo. Don't look too closely, these elements kinda
suck right now.
* ext/cairo/gstcairo.c: new
* ext/cairo/gsttextoverlay.c: new
* ext/cairo/gsttextoverlay.h: new
* ext/cairo/gsttimeoverlay.c: new
* ext/cairo/gsttimeoverlay.h: new
* gst-libs/gst/media-info/media-info-priv.h: fix compile
problem with compilers that don't support variadic macros.
Original commit message from CVS:
* gst/asfdemux/README
* gst/wavenc/riff.h
* gst-libs/gst/riff/riff-ids.h
* gst-libs/gst/riff/riff-media.c
add new 4CC codes for h263 related codecs
fixes partially bug #155163
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_chanpos_from_gst),
(gst_faad_chanpos_to_gst), (gst_faad_chain):
Set DURATION even if source buffer didn't. Also use increasing
timestamps.
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_audio_caps_with_data):
Block_align can have larger values than 8192.
Original commit message from CVS:
* ext/esd/esdsink.c: (gst_esdsink_chain):
Make error actually say something useful (fixes#156798).
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data),
(gst_riff_create_video_template_caps):
Add Intel Video 5.0 fourcc (IV50).
Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_use_event):
Don't forward DISCONT events (fixes#159684).
Original commit message from CVS:
2004-11-27 Martin Soto <martinsoto@users.sourceforge.net>
* gst-libs/gst/audio/audioclock.c (gst_audio_clock_set_active)
(gst_audio_clock_get_internal_time):
Fix active <-> inactive transitions: ensure time value always
grows and avoid abrupt value changes.
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_get_caps_internal):
buffer-frames property was missing
* ext/arts/gst_arts.c:
rate missing from sinkcaps
* ext/audiofile/gstafparse.c:
* ext/audiofile/gstafsink.c:
* ext/audiofile/gstafsrc.c:
* ext/swfdec/gstswfdec.c:
int audio doesn't know buffer-frames
* ext/cdparanoia/gstcdparanoia.c:
int audio doesn't know chunksize either
* ext/nas/nassink.c:
it's endianness, not endianess
* gst-libs/gst/audio/audio.h:
make float standard pad template caps really describe float
* gst/law/mulaw.c: (linear_factory):
signed only, please
* gst/mpegstream/gstdvddemux.c:
widths of 20 are not valid
