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.
