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
