Allows finer grain decisions about formats and features at each
stage of the pipeline.
Also provide propose_allocation for glupload besed on the supported
methods.
Make GstGLMemory hold the texture target (tex_target) the texture it represents
(tex_id) is bound to. Modify gst_gl_memory_wrapped_texture and
gst_gl_download_perform_with_data to take the texture target as an argument.
This change is needed to support wrapping textures created outside libgstgl,
which might be bound to a target other than GL_TEXTURE_2D. For example on OSX
textures coming from VideoToolbox have target GL_TEXTURE_RECTANGLE.
With this change we still keep (and sometimes imply) GL_TEXTURE_2D as the
target of textures created with libgstgl.
API: modify GstGLMemory
API: modify gst_gl_memory_wrapped_texture
API: gst_gl_download_perform_with_data
Depending on the platform, it was only ever implemented to 1) set a
default surface size, 2) resize based on the video frame or 3) nothing.
Instead, provide a set_preferred_size () that elements/applications
can use to request a certain size which may be ignored for
videooverlay/other cases.
A context can create a GLsync object that can be waited on in order
to ensure that GL resources created in one context are able to be
used in another shared context without any chance of reading invalid
data.
This meta would be placed on buffers that are known to cross from
one context to another. The receiving element would then wait
on the sync object to ensure that the data to be used is complete.
This gives more flexibility to the subclasses and permits to remove the
GstVideoAggregatorClass->disable_frame_conversion ugly API.
WARNING: This breaks the API as it removes the disable_frame_conversion
field
API:
+ GstVideoAggregatorClass->find_best_format
+ GstVideoAggregatorPadClass->set_format
+ GstVideoAggregatorPadClass->prepare_frame
+ GstVideoAggregatorPadClass->clean_frame
- GstVideoAggregatorClass->disable_frame_conversion
https://bugzilla.gnome.org/show_bug.cgi?id=740768
If we seek when media is in stop state, playback-test gives
critical error, since context of glimagesink is destroyed during stop.
But since context is not present, we need not handle send_event in glimagesink
Hence adding a condition to check if context is valid.
https://bugzilla.gnome.org/show_bug.cgi?id=740305
gst_glimage_sink_handle_events can be called from the overlay interface and from
the main thread before GL is setup. Before this change, that would call
_ensure_gl_setup() and deadlock on OSX.
Change things so that it's always safe to call gst_glimage_sink_handle_events()
without stuff deadlocking.
Remove gst_glimage_sink_handle_events call in gst_glimage_sink_init. It was
unnecessary and when the element was instantiated from the main thread, caused a
deadlock in OSX creating the context (thread).
with force-aspect-ratio=true, if the width or height changed, the
viewport wasn't being updated to respect the new video width and height
until a resize occured.
Otherwise, it is only possible for the sink pads and the src pads to
have the exact same caps features. We can convert from any feature
to another feature so support that.
Otherwise, it is only possible for the sink pads and the src pads to
have the exact same caps features. We can convert from any feature
to another feature so support that.
- update for shaders
- add alpha property
- image placement properties shamelessly borrowed from gdkpixbufoverlay
- image placement properties are GstController able
- use GstGLMemory for the overlay image data
- add support for gles2
Otherwise we could pass on a RGBA formatted buffer and downstream would
misinterpret that as some other video format.
Fixes pipelines of the form
gleffects ! tee ! xvimagesink
Allows callers to properly reference count the buffers used for
rendering.
Fixes a redraw race in glimagesink where the previous buffer
(the one used for redraw operations) is freed as soon as the next
buffer is uploaded.
1. glimagesink uploads in _prepare() to texture n
1.1 glupload holds buffer n
2. glimagesink _render()s texture n
3. glimagesink uploads texture n+1
3.1 glupload free previous buffer which deletes texture n
3.2 glupload holds buffer n+1
4. glwindow resize/expose
5. glimagesink redraws with texture n
The race is that the buffer n (the one used for redrawing) is freed as soon as
the buffer n+1 arrives. There could be any amount of time and number of
redraws between this event and when buffer n+1 is actually rendered and thus
replaces buffer n as the redraw source.
https://bugzilla.gnome.org/show_bug.cgi?id=736740
* aspect should not be 0 on init
* rename fovy to fov
* add mvp to properties as boxed graphene type
* fix transformation order. scale first
* clear color with 1.0 alpha
https://bugzilla.gnome.org/show_bug.cgi?id=734223
Dynamic pipelines that get and release the sink pads will finalize
the pad without going through gst_gl_mixer_stop() which is where the
upload object is usually freed. Don't leak objects in such case.
If window is resized, GstStructure pointer values have to be rescaled to
original geometry. A get_surface_dimensions GLWindow class method is added for
this purpose and used in the navigation send_event function.
https://bugzilla.gnome.org/show_bug.cgi?id=703486
The expected default behaviour for video sink is to maintain the
aspect ratio. Fix the default value to reflect this. The property
default was already TRUE, but the value was not initially TRUE.
Allows automatic negotiation of the size in the following case:
gst-launch-1.0 glvideomixer name=m sink_0::xpos=0 sink_1::xpos=320 ! glimagesink \
videotestsrc ! m. \
videotestsrc pattern=1 ! m.
https://bugzilla.gnome.org/show_bug.cgi?id=731878
This is too allow gst-launch debugging with multiple GL contexts as
well as avoiding segfaulting innocent gtk+ apps that have not called
XInitThreads.
https://bugzilla.gnome.org/show_bug.cgi?id=731525
The reshape property was never used.
Replace the draw property with a signal.
Based on patch by Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>
https://bugzilla.gnome.org/show_bug.cgi?id=704507
This patch provides the basic infrastructure required for this.
Upload and Download has been ported to this.
Has the nice effect of allowing GstGLMemory to be our
refcounted texture object for any texture type (not just RGBA).
Should not lose any features/video formats.
gst_gl_context_create() might need to dispatch some operations to the
application's main thread, and calling this in the change_state function
can cause deadlocks.
* picked from old libgstegl:
- GstEGLImageMemory
- GstEGLImageAllocator
- last_buffer management from removed GstEGLImageBufferPool
* add-ons:
- GstEGLImageMemory now old a reference on GstGLContext
so that it can delete the EGLImage and its gltexture source
while having the associated gl context being current.
- add EGLImage support for GstVideoGLTextureUploadMeta which
mainly call EGLImageTargetTexture2D
- GstGLBufferPool now supports GstEGLImageAllocator
- glimagesink / glfilters / etc.. now propose GstEGLImageAllocator
to upstream
https://bugzilla.gnome.org/show_bug.cgi?id=703343
Make buffer timestamps more accurate and, more importantly, actually
representative of the MIDI events timing.
Previously, buffers were only sent with timetamps aligned at a 10ms
boundary which was just wrong, now the buffer timestamp represents the
real time of the MIDI event.
Conveniently, the ALSA sequencer API supports scheduling events in the
future so the sequencer infrastructure can be used to have the tick
delivered at the right time, avoiding any custom scheduling mechanism.
The ticks scheduling starts on the first transition to PLAYING, and the
delay is also calculated when the pipeline goes into PLAYING.
https://bugzilla.gnome.org/show_bug.cgi?id=787683
When setting the "ports" property the value is duplicated but it's not
freed when the elements stops.
Reported by Valgrind (example run with "alsamidisrc ports=128:0"):
6 bytes in 1 blocks are definitely lost in loss record 30 of 1,911
at 0x4C2BBEF: malloc (in /usr/lib/valgrind/vgpreload_memcheck-amd64-linux.so)
by 0x5411528: g_malloc (gmem.c:94)
by 0x542A9FE: g_strdup (gstrfuncs.c:363)
by 0x775211E: gst_alsa_midi_src_set_property (gstalsamidisrc.c:284)
by 0x5184A4D: object_set_property (gobject.c:1439)
by 0x5184A4D: g_object_setv (gobject.c:2245)
by 0x51859DD: g_object_set_property (gobject.c:2529)
by 0x4F0474C: ??? (in /usr/lib/x86_64-linux-gnu/libgstreamer-1.0.so.0.1203.0)
by 0x4F065C8: ??? (in /usr/lib/x86_64-linux-gnu/libgstreamer-1.0.so.0.1203.0)
by 0x4F07557: ??? (in /usr/lib/x86_64-linux-gnu/libgstreamer-1.0.so.0.1203.0)
by 0x4EFE3EE: gst_parse_launch_full (in /usr/lib/x86_64-linux-gnu/libgstreamer-1.0.so.0.1203.0)
by 0x4EFE673: gst_parse_launchv_full (in /usr/lib/x86_64-linux-gnu/libgstreamer-1.0.so.0.1203.0)
https://bugzilla.gnome.org/show_bug.cgi?id=787683
We only need to initialize the mutex/cond once when creating the
element and then release them when we are done with the element.
Avoids weird "mutex_clear called when still locked" issues
There were still some races going on where seeking events wouldn't
be properly intercepted/executed by this thread.
* Instead of always waiting for the GCond to be emitted, first just
check if there is an event available
* Take ownership of the event *while* the lock is taken and not
after releasing/reacquiring it
* Finally acquire lock at the very top and release it at the end
to make it a bit more streamlined
This removes the remaining issues with seeks not being executed
The previous branch will release the lock in the call to
gst_ogg_demux_seek_back_after_push_duration_check_unlock()
Only unlock it if we didn't call that function
When calculating duration in push-mode we seek to a certain position
and discard any data until we get data from that requested position.
The problem is that basing ourselves solely on offset to determine
whether we reached the target offset is wrong since the source might
be fast enough to send us that target position *before* it processed
the requested seek.
This would end up in a situation where:
* We think we're done with duration estimate
* We fire a seek back to "0" in the loop thread
* We resume normal processing
* ... except that we're still getting data from too far ahead which
we decide to process.
* And we start doing totally wrong granule/time/duration calculation
and pushing wrong data.
Instead of this confusion, wait until we receive data from the requested
seek. We do that by using the fact that the seqnum in
seek_event_drop_til will be non-zero until the SEGMENT corresponding
to the requested SEEK has been received.
Bonus: makes startup slightly faster
Code using the push_loop_thread (using for sending seeks) assumes
that the thread was properly started, except that this isn't always
true and the thread might not have completely started.
Instead wait for the thread to properly start before doing anything
else.
If we are going to return a (potentially) 64bit integer, don't use
a 32bit one for calculation, otherwise we could end up exceeding
the maximum size of a 32bit int.
For stream mappers that don't set a specific granuleshift, it will
have the default value of -1.
Protect the code for that and return the granule value as-is
Since the default value of a GstOggPad.map.map was 0 ... we would
end up using wrong functions from mappers() if the stream wasn't
initialized yet.
Instead of that, use a default blank/empty first entry.
In some corner cases we end up with the building chain not being
properly tracked (and therefore not properly freed).
Add a FIXME so it can later be fixed, but for now just fix the leak
... as expected later on when end time is used to determine end running time.
Otherwise the latter is determined as NONE and the resulting text buffer is
then used indefinitely.
Fix various issues with reverse playback by clearing tracking
vars when working in reverse, and where possible using the
timestamp interpolation code to generate timestamps for
outgoing buffers. Make sure to mark things as discontinuous
only when looping backward to a new position and fix seeking
to the next page when starting.
In gst_ogg_demux_do_seek() when calculating the
keyframe time, account for a non-zero start-time
Handle a discontinuous first packet in
gst_ogg_demux_setup_first_granule() because that's pretty
normal after a seek. Also differentiate between a genuinely
truncated first packet and just bailing out early, by not using
granule = -1 as an error code.
Make the debug output logs clearer about which timestamps
are stream times (PTS) and which are ogg timestamps.
This is a followup commit to b95725c37e
* Resetting the decoder should only happen when we get a new initialization
header (0x01) and not on the other headers
* The initialized variable only gets set to TRUE once all headers have
been parsed. Also check if the vorbis_info struct has been properly resetted
also. Failure to do that would cause vorbisdec to error if it got
two initialization header in a row (the first would configure the underlying
library and the second one would error out because it's already initialized)
https://bugzilla.gnome.org/show_bug.cgi?id=779515
This fixes missing audio when we get buffers with zero
duration, denoting unknown duration. When several such
buffers are received in a row, they're all at the same
timestamp, with zero duration.
https://bugzilla.gnome.org/show_bug.cgi?id=771723
Always supply a buffer with max size to the decoder, as we
can't really decide how many samples will be in the lost packet
based on the timestamps we get.
https://bugzilla.gnome.org/show_bug.cgi?id=771723
If we can't find a valid granule near the end of the file, we
disable seeking. This guards against the whole file being then
read and never going to PLAYING.
https://bugzilla.gnome.org/show_bug.cgi?id=770314
https://github.com/mesonbuild/meson
With contributions from:
Tim-Philipp Müller <tim@centricular.com>
Jussi Pakkanen <jpakkane@gmail.com> (original port)
Highlights of the features provided are:
* Faster builds on Linux (~40-50% faster)
* The ability to build with MSVC on Windows
* Generate Visual Studio project files
* Generate XCode project files
* Much faster builds on Windows (on-par with Linux)
* Seriously fast configure and building on embedded
... and many more. For more details see:
http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.htmlhttp://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html
Building with Meson should work on both Linux and Windows, but may
need a few more tweaks on other operating systems.
This workaround tried to avoid an EOS event when seeking to the
end of an Ogg stream in order to find its duration. At some point,
an EOS event there would cause any queue2 upstream to pause and
not restart on a seek back to the beginning. This now appears to
not be the case anymore, and so the workaround can be removed.
https://bugzilla.gnome.org/show_bug.cgi?id=767689
This reverts commit a16cd5d2a5.
Setting the stop time on the segment breaks reconfiguration, as the
encoder signals an EOS, but we reconfigure it an continue to produce
buffers.
This information should not be required via the segment downstream
since we already have the sample count being used to generate buffer
durations.
https://bugzilla.gnome.org/show_bug.cgi?id=768763
If the duration is not known from the chain, it might be known
by the startup seek.
This fixes failure to seek.
Merged with a patch from Tim-Philipp Müller <tim@centricular.com>
https://bugzilla.gnome.org/show_bug.cgi?id=768991
Dropping a buffer because we have a seek pending is normal,
and will now happen when we trigger a seek while going through
the packets in a page. So this should not be an error.
A low bitrate stream which can pack more than 2 seconds of audio
in a page would cause the stream's position to be updated not
often enough, and would trigger a spurious "jump" via a GAP
event. Instead, we update the stream position after calculating
the new overall segment position.
https://bugzilla.gnome.org/show_bug.cgi?id=764966
The only way for ALSA to expose a position-less multi channels is to
return an array full of SND_CHMAP_MONO. Converting this to a
GST_AUDIO_CHANNEL_POSITION_MONO array would be invalid as
GST_AUDIO_CHANNEL_POSITION_MONO is meant to be used only with one
channel.
Fix this by using GST_AUDIO_CHANNEL_POSITION_NONE which is meant to be
used for position-less channels.
https://bugzilla.gnome.org/show_bug.cgi?id=763799
Introduces [x-absolute, y-absolute] properties
for positioning in +/- MAX_DOUBLE range.
Adds new (h/v)alignment type "absolute" where coordinates
map the text area to be exactly inside of video canvas for [0, 0] - [1, 1]:
[0, 0]: Top-Lefts of video and text are aligned
[0.5, 0.5]: Centers are aligned
[1, 1]: Bottom-Rights are aligned
https://bugzilla.gnome.org/show_bug.cgi?id=761251
libgstreamer currently exports some debug category
symbols GST_CAT_*, but those are not declared in any
public headers.
Some plugins and libgstvideo just use GST_DEBUG_CATEGORY_EXTERN()
to declare and use those, but that's just not right at
all, and it won't work on Windows with MSVC. Instead look
up the categories via the API.
In order to detect graphical user input on the
textoverlay, the resulting rendering properties
need to be exposed to applications.
Fixes delayx property declaration.
https://bugzilla.gnome.org/show_bug.cgi?id=761251
The current position property is limited to X,Y positions
in the range of [0, 1]. This patch allows full control
over the overlay position, including partially outside
of the video area.
https://bugzilla.gnome.org/show_bug.cgi?id=761251
FEC may only be used when PLC is enabled on the audio decoder,
as it relies on empty buffers to generate audio from the next
buffer. Hooking to the gap events doesn't work as the audio
decoder does not like more buffers output than it sends.
The length of data to generate using FEC from the next packet
is determined by rounding the gap duration to nearest. This
ensures that duration imprecision does not cause quantization
to 2.5 milliseconds less than available. Doing so causes the
Opus API to fail decoding. Such duration imprecision is common
in live cases.
The buffer to consider when determining the length of audio
to be decoded is the previous buffer when using FEC, and the
new buffer otherwise. In the FEC case, this means we determine
the amount of audio from the previous buffer, whether it was
missing or not (and get the data either from this buffer, or
the current one if the previous one was missing).
rename gst-launch --> gst-launch-1.0
replace old elements with new elements(ffmpegcolorspace -> videoconvert, ffenc_** -> avenc_**)
fix caps in examples
https://bugzilla.gnome.org/show_bug.cgi?id=759432
The result of the two expressions will be promoted to guint64 anyway,
perform all the arithmetic in 64 bits to avoid potential overflows.
CID 1338690, CID 1338691
We always require the channel-mapping-field. If it's 0 we require nothing
else, otherwise we need channels, stream-count and coupled count to be
available.
oggdemux is outputting the meta now, and only outputs if it should really
apply to the current buffer. Previously we would skip N samples also if we
started the decoder in the middle of the stream.
https://bugzilla.gnome.org/show_bug.cgi?id=757153
It is doing the wrong thing because of the Opus pre-skip: while the timestamps
are shifted by the pre-skip, the granule positions are not shifted.
oggmux is doing the right thing here already.
https://bugzilla.gnome.org/show_bug.cgi?id=757153
The first frame has lookahead less samples, the last frame might have some
padding or we might have to encode another frame of silence to get all our
input into the encoded data.
This is because of a) the lookahead at the beginning of the encoding, which
shifts all data by that amount of samples and b) the padding needed to fill
the very last frame completely.
Ideally we would use LPC to calculate something better than silence for the
padding to make the encoding as smooth as possible.
With this we get exactly the same amount of samples again in an
opusenc ! opusdec pipeline.
https://bugzilla.gnome.org/show_bug.cgi?id=757153
... instead of relying on the segment. For the clipping at the start we assume
a proper value in the OpusHead, as generated by opusparse or opusenc.
Transmuxing in general is not guaranteed to produce the correct values, or
even have a OpusHead (e.g. when having RTP input).
https://bugzilla.gnome.org/show_bug.cgi?id=757153
The granulepos does not have the pre-skip subtracted while timestamps do,
and the last granulepos will be shorter by the number of samples that should
be dropped because of padding in the end.
As such, extrapolating the granule of the beginning of the first frame will
lead to a negative value, which is not a problem but intentional.
https://bugzilla.gnome.org/show_bug.cgi?id=757153
Usually these loops only run once, so there's no problem here. But sometimes
they run twice, and by adding the number of bytes to a 16 bit pointer type we
would advance twice as much as we should.
Also use snd_pcm_frames_to_bytes() in alsasrc to calculate
the number of bytes to skip, same as we do in alsasink.
Thanks to Lucio A. Hernandez <lucio.a.hernandez@gmail.com> for reporting.
The alsamidisrc element allows to get input event from ALSA MIDI
sequencer devices, and possibly convert them to sound using some
downstream element like fluiddec.
Fixes#738687
It is faster than doing a query that propagates downstream and
should be enough
Elements: faac, gsmenc, opusenc, sbcenc, voamrwbenc, adpcmenc, sirenenc
Removes the need for custom caps query handling and makes it more
correct from the beginning on the template. It is a bit uglier
to read because there is 1 entry per channel but makes code easier
to maintain.
accept-caps is not recursive and might stop at the next downstream element,
while caps queries are generally recursive. The next element might accept any
capsfeatures we want, but that doesn't mean that further downstream it will
also work.
Additionally for the future:
We should probably check if downstream *prefers* the
overlay meta, and only enforce usage of it if we can't handle
the format ourselves and thus would have to drop the overlays.
Otherwise we should prefer what downstream wants here.
the extents rectangle is what you need to know to properly position
a buffer that has been rendered in a surface of the ink rectangle
size. This patch make the placement on par with the placement we had
before without having to over allocate.
This patch also enable placement for vertical rendering. Note that
the halginement, valighment and line-alignment default are set to
the previous default when this property is set. This is for backward
compatibility, you can change the value after setting vertical render.
https://bugzilla.gnome.org/show_bug.cgi?id=728636
This patch uses the ink rectangle in order to compute the size
of the surface require to render. It also correctly compute the
transformation matrix as the ink_rect position might not be at
0, 0. Additionally, shadow_offset and outline_offset (which is
in fact the diameter of a dot, not a really an offset) is now
taken into account. Redundant matrix operation has been removed
for the vertical rendering.
Take note that the matrix operation in cairo are excuted in
reverse order.
https://bugzilla.gnome.org/show_bug.cgi?id=728636
* Only send the caps event once if the query had support for the
overlay composition meta.
* Only do the allocation query if it is supported through caps.
* Send overlay_caps before doing allocation query rather then normal
caps
https://bugzilla.gnome.org/show_bug.cgi?id=751157
The GstVideoOverlayComposition meta coordinates should always be
in stream scale, regardless of the window size downstream. This
way the sink can always scale the composition if the window size
have changed after a buffer (with his meta) was rendered before.
https://bugzilla.gnome.org/show_bug.cgi?id=751157
This avoids negotiating twice. Current the _setcaps() patch does
not clear the initial reconfigure flags, which lead to systematic
double renegotiation.
http://bugzilla.gnome.org/show_bug.cgi?id=751157
Remove the optimization to skip allocation query so we can
always have the latest window size information. Also, correctly
deal with the case where there is no window size information.
http://bugzilla.gnome.org/show_bug.cgi?id=751157
* cache window size event and update handle ratio
* init width with 1, don't use 0
* don't update overlay when receiving same window size
* receive window size from allocation query
https://bugzilla.gnome.org/show_bug.cgi?id=751157
This makes pipelines with multiple textoverlay elements possible.
The meta data is collected from the upstream textoverlay element,
merged into a new GstVideoOverlayComposition and passed down downstream.
https://bugzilla.gnome.org/show_bug.cgi?id=751157
Previously, PLC frames always had a length of 120ms, which caused audio
quality degradation and synchronization errors. Fix this by calculating an
appropriate length for the PLC frame.
The length must be a multiple of 2.5ms. Calculate a multiple of 2.5ms that
is nearest to the current PLC length. Any leftover PLC length that didn't
make it into this frame is accumulated for the next PLC frame.
https://bugzilla.gnome.org/show_bug.cgi?id=725167
Have all sections in alphabetical order. Also make the macro order consistent.
This is a preparation for generating the file. Remove GET_CLASS macro for
some elements, since it is not used and the header is not installed.
The intention was to skip the allocation query if upstream has decided
to use the overlay meta feature in the caps. We can safely assume that
upstream have done that query already before making this decision. This
is an optimization since doing allocation queries is relatively
expensive.
CID #1308943
upstream_has_meta is set to FALSE and never changed. The two checks for if
upstream_has_meta will never go to the true branch. Removing the boolean
and the true branches of these checks.
CID #1308943
This cleanup the negotiation function by properly splitting the probe
and the decisions. This allow handling correctly pipeline where upstream
caps have special memory type. An example pipeline is:
gltestsrc ! textoverlay text=bla ! fakesink
The upstream caps will be memory:GLMemory, which isn't supported by the
blitter.
https://bugzilla.gnome.org/show_bug.cgi?id=749243
This reverts commit 76647f2710.
Avoiding pull mode activation is a feature regression, and
demuxers should always use pull mode where that is possible,
e.g. if there's an upstream queue2 with a ring buffer or
a download buffer.
This patch made reverse playback no longer possible over http.
If the goal is to minimise seeks, then that can still be done
by making the demuxer behave differently in pull mode if
the SEQUENTIAL flag is set. If there are bugs, like the demuxer
needlessly scanning the entire file on start-up in pull mode,
then those should be fixed instead.
https://bugzilla.gnome.org/show_bug.cgi?id=746010
gst_event_replace() takes its own reference on the event so we should drop
ours after creating and storing an event using it.
This fix leaks which can be reproduced using the
validate.http.media_check.vorbis_theora_1_ogg scenario.
https://bugzilla.gnome.org/show_bug.cgi?id=748247
If we don't consume the buffer by passing its reference to
overlay->text_buffer then we need to unref it.
Fix a leak with validate.file.playback.fast_forward.test5_mkv
when running inside Valgrind.
https://bugzilla.gnome.org/show_bug.cgi?id=747602
When a stream has a skeleton index, the stream time is taken from that
index. However, when part of the stream is captured, the index is
invalid as its offsets are now wrong. To avoid this, we ignore the index
when the last offset points beyond the end of the stream (when its
byte length is known).
https://bugzilla.gnome.org/show_bug.cgi?id=744070
When deltax is large enough to cause the text to push past the
width of the frame, it would disappear due to a bug in setting
the layout width.
While there, fix a log printing an incorrect width to set.
https://bugzilla.gnome.org/show_bug.cgi?id=739689
oggmux keeps a cached buffer per pad, and pulls buffers from
collectpads to this cached buffer for all pads before processing
the best pad. In some cases, the move from collectpads buffer
to cached buffer is delayed till next call. However, when there
is only one pad, this can't be delayed till next call as there
will be a deadlock since collectpads has no other pad to push to.
https://bugzilla.gnome.org/show_bug.cgi?id=740565
This way we let opusdec do the resampling if needed and don't carry
around buffers with a too high sample rate if not required.
While Opus always uses 48kHz internally, this information from the
header specifies which frequencies are safe to drop.
gstoggdemux.c:1233:11: error: format specifies type 'long' but the argument has type 'ogg_int64_t' (aka 'long long') [-Werror,-Wformat]
granule);
^~~~~~~
https://bugzilla.gnome.org/show_bug.cgi?id=746512
The code that was calculating the start granule from packet durations
was interpreting a negative value as an error, but this is actually a
valid case, to indicate clipping of data at start.
https://bugzilla.gnome.org/show_bug.cgi?id=743900
If we get EOS when we're trying to build a chain, we disable seeking
and continue instead of posting an error. This can happen for corner
cases such as a stream with a video that stops before the end, for
instance.
https://bugzilla.gnome.org/show_bug.cgi?id=745980
When looking for pages when seeking, we stop looking for non sparse
streams if we don't find one within a given threshold. This fixes
seeking filling up queues and blocking in corner cases such as an
audio file with a pathological 1 frame video stream (yes, I saw one).
https://bugzilla.gnome.org/show_bug.cgi?id=745980
Store the video info of the internal frame decode width/height
separate to the exposed (cropped) frame info, so that it can be
used for mapping the downstream allocated video frame buffer correctly
when using GstVideoCropMeta.
Fixes playback of files with sizes that aren't a multiple of 16-pixels
width or height.
https://bugzilla.gnome.org/show_bug.cgi?id=741030
This will usually deadlock, despite this patch being in master for
quite some time and working fine. Nevertheless, we deem it to be
not working, disregarding facts.
As such, we fix it by keeping track of seek events, and sending
them upstream from a separate thread. Buffers are then discarded
till we get a new segment with the expected seqnum.
READY->PAUSED can be too early as souphttpsrc can get the HTTP
headers after this. Try again in the chain function.
Also use seeking query to disable seeking if upstream reports
being unseekable.
Some resetting code has to be done in the NEW_SEGMENT
event handler, instead of the missing FLUSH_STOP one.
Segment base was also wrongly accounted for. This was hidden
by the fact that flushing resets the base.
A discontinuity is now also signalled on seeking. We have to
also ensure that the discontinuity "sticks" till a buffer
with a valid timestamp goes out, or the audio decoder base
class will ignore the discontinuity for purposes of keeping
track of the current time.
This allows using non flushing segment seeks for looping
HTML audio in particular, and more generally non flushing seeks.
https://bugzilla.gnome.org/show_bug.cgi?id=729198
The code was using the first nonnegative granulepos to seed the
granule tracking, which appeared to work since headers have zero
granulepos. However, this does not work for files with a hole at
start, which are common in live streaming.
The correct behavior is to look for the first granule, and subtract
the duration of all the packets finishing on this page.
The function which does this relies on the fact that the ogg_stream
structure can be duplicated by shallow copy, in order to pull the
packets from the first page(s) on the copy without affecting the
original stream state.
The max latency parameter is "the maximum time an element
synchronizing to the clock is allowed to wait for receiving all
data for the current running time" (docs/design/part-latency.txt).
https://bugzilla.gnome.org/show_bug.cgi?id=744338
Don't use private GMutex implementation details to check
whether it has been freed already or not. Just turn dispose
function into finalize function which will only be called
once, that way we can just clear the mutex unconditionally.
Makes theora work in cases where the header packets are only in the caps
(because theoradec was connected to oggdemux late and missed the
beginning of the stream)
If the streaming task attempts to read a chain while the pipeline
is stopping (which can happen if the pipeline stops shortly after
start or a new URI being setup in gapless playback case), it will
see a flushing return from upstream, and should then also return
flushing to the caller, rather than emit a flow error.
https://bugzilla.gnome.org/show_bug.cgi?id=722442
Fix leak of caps event and of caps objects when setting caps on sink and src
pads. Sync audiovisualizer class implementation to the one in gst-plugins-bad.
This commit matches c5ef1bee73 in that module.
https://bugzilla.gnome.org/show_bug.cgi?id=742875
This reverts commit a91d521a36.
Being a base class it is better to check the value instead of ignoring it since
a child class could be created that returns valuable information.
klass->setup (scope) will always return TRUE since all children of this class
do so, no need to store the return. Besides, the value is overwritten a few
lines down before it is ever used. Save the unnecessary memory and instructions.
CID #1226467
ret is assigned but not used and in the next cycle of the loop it is overwritten
with default_prepare_output_buffer (). If there is a flow error the function
should return instead.
CID #1226475
It might happen that the timestamp is before the segment and the
check would succeed. In this case reducing the duration makes no
sense and would lead to broken results.
The previous code was setting keytarget to target
to make sure the keyframe found for each pad was
indeed before the target.
Then if target == keytarget, it assumed a keyframe had been
found, which was not the case if target was before the first frame
in the file.
This patch checks that a keyframe was indeed found, and if not
seeks to 0, without bisecting again.
Assuming default gst qa assets in $HOME/gst-validate
seek_before_first_frame.scenario:
description, seek=true, handles-states=true
pause, playback-time=0.0
seek, playback-time=0.0, start=0.0, flags=accurate+flush
seek, playback-time=0.0, start=0.01, flags=accurate+flush
seek, playback-time=0.0, start=0.1, flags=accurate+flush
GST_DEBUG=*theoradec*:2 gst-validate-1.0 playbin \
uri=file://$HOME/gst-validate/gst-qa-assets/medias/ogg/vorbis_theora.0.ogg \
--set-scenario seek_before_first_frame.scenario
https://bugzilla.gnome.org/show_bug.cgi?id=741097
When encoding, libvorbis will tell us how many samples are encoded
in the buffer it returns. This number may be less than the maximum
of samples in the block, if this is the last packet. In we have no
segment end time, we set it to the end time of that last sample to
tell downstream that the buffer contains less samples.
Samples may be clipped at the end, and this is conveyed by a
granulepos that's smaller than it would otherwise be. Use the
segment stop time to detect this, and calculate the right
granulepos.
When the textoverlay is set outside the video frame by deltax or deltay the
calculation segfaults, but it is also unnecessary since it doesn't need to be
displayed. So we should clip the text.
https://bugzilla.gnome.org/show_bug.cgi?id=738242
vorbis_reorder_map is defined for eight channels max. If we have more
than eight channels, it's the application which shall define the order.
Since we set audio position to none, we just interleave all the channels
without any particular reordering.
https://bugzilla.gnome.org/show_bug.cgi?id=737742
The allocation query failure doesn't mean that the negotiation
has failed as the element can allocate buffers itself.
Instead, only fail if the pads are flushing and the allocation
query failed.
https://bugzilla.gnome.org/show_bug.cgi?id=735844
The source pad might be flushing while negotiating, resulting in
set_caps or the ALLOCATION query failing. In this case set the
reconfigure flag on the source pad so that negotiation is retried on the
next buffer.
When downstream claims to accept the overlay meta but fails to
provide it in the allocation query, properly fallback to setting
a new caps without the overlay meta as that is not going to be used.
Only do this if the original caps doesn't have the overlay already,
otherwise there isn't much that can be done.
https://bugzilla.gnome.org/show_bug.cgi?id=735800
Setting segment.base in the segment sent from gst_ogg_demux_handle_page() is
enough to ensure that chained oggs are played corretly (see bgo#706569).
Tweaking the base in gst_ogg_pad_submit_packet() as well result in delays when
playing a file with start != -1.
https://bugzilla.gnome.org/show_bug.cgi?id=735808
It's not done in any other code calling negotiate and will cause deadlocks
as it is sending events and queries in the pipeline.
Specifically this pipeline was deadlocking:
gst-launch-1.0 videotestsrc ! textoverlay ! textoverlay ! fakesink
Base time should be accumulated so non flushing seeks have the expected base.
Not accumulating result in segments appearing as "too late" and so are not
played by the sink.
https://bugzilla.gnome.org/show_bug.cgi?id=735509
Make textoverlay negotiate caps more correctly.
1) Check what caps we received in the video-sink
2) If it already has the overlay meta -> use it directly
3) If it doesn't, textoverlay try adding the overlay meta and using it,
if downstream doesn't support it, just use what is received in the
video-sink
4) Check if the allocation query also supports the meta to enable
really using it
Before it wasn't really doing renegotiation of any kind, just
re-checking if it should use the overlay meta or not
Also had to update the caps in the test as memory:SystemMemory seems
to be required when you use a caps feature otherwise intersection/subset
checks will fail.
https://bugzilla.gnome.org/show_bug.cgi?id=733916
The headers were never getting reffed when being added to the headers
list, which is later unreffed-and-freed by the caller (e.g.
gst_opus_parse_parse_frame()).
https://bugzilla.gnome.org/show_bug.cgi?id=733013
When a pipeline using alsasink and push mode upstream fails
to preroll, the following state will be the case:
- A loop upstream will be PAUSED, pushing a first buffer
- alsasink will be READY, pending PAUSED, because async
On error, the pipeline will switch to NULL. alsasink is in
READY, so goes to NULL immediately. It zeroes its cached
caps. Meanwhile, the upstream loop can cause a caps query,
conccurent with the state change. This will use those cached
caps. If the zeroing happens between the NULL test and the
dereferencing, GStreamer will critical down in the GstValue
code.
Since it appears that such a gap between states (PAUSED
and pushing upstream, and NULL downstream) is expected, we
need to protect the read/write access to the cached caps.
This fixes the critical.
See https://bugzilla.gnome.org/show_bug.cgi?id=731121
This lets oggdemux determine they are not delta units, and removes
spurious per packet warnings about being unable to determine the
packet's keyframeness.
They are very confusing for people, and more often than not
also just not very accurate. Seeing 'last reviewed: 2005' in
your docs is not very confidence-inspiring. Let's just remove
those comments.
This should not cause any actual bug since Theora and Daala have
a maximum shift of 31, and a packet duration of 2^31 seems very
implausible. But it fixes:
Coverity 1139804, 1139803, 1139802
