This allows waylandsink to fail gracefully before going to READY
in case one of the required interfaces does not exist. Not all
interfaces are necessary for all modes of operation, but it is
better imho to fail before going to READY if at least one feature
is not supported, than to fail and/or crash at some later point.
In the future we may want to relax this restriction and allow certain
interfaces not to be present under certain circumstances, for example
if there is an alternative similar interface available (for instance,
xdg_shell instead of wl_shell), but for now let's require them all.
Weston supports them all, which is enough for us now. Other compositors
should really implement them if they don't already. I don't like the
idea of supporting many different compositors with different sets of
interfaces implemented. wl_subcompositor, wl_shm and wl_scaler are
really essential for having a nice video sink. Enough said.
This essentially hides the video and allows the application to
potentially draw a black background or whatever else it wants.
This allows to differentiate the "paused" and "stopped" modes
from the user's point of view.
Also reworded a comment there to make my thinking more clear,
since the "reason for keeping the display around" is not really
the exposed() calls, as there is no buffer shown in READY/NULL
anymore.
1) We know that gst_wayland_sink_render() will commit the surface
in the same thread a little later, as gst_wl_window_set_video_info()
is always called from there, so we can save the compositor from
some extra calculations.
2) We should not commit a resize with the new video info while we are still
showing the buffer of the previous video, with the old caps, as that
would probably be a visible resize glitch.
Previously, in order to change the surface size we had to let the pipeline
redraw it, which at first also involved re-negotiating caps, etc, so a
synchronization with the pipeline was absolutely necessary.
At the moment, we are using wl_viewport, which separates the surface size
from the buffer size and it also allows us to commit a surface resize without
attaching a new buffer, so it is enough to just do:
gst_wayland_video_pause_rendering():
wl_subsurface_set_sync()
gst_video_overlay_set_render_rectangle():
wl_subsurface_set_position()
wl_viewport_set_destination()
wl_surface_damage()
wl_surface_commit()
... commit the parent surface ...
gst_wayland_video_resume_rendering():
wl_subsurface_set_desync()
This is enough to synchronize a surface resize and the pipeline can continue
drawing independently. Now of course, the names pause/resume_rendering are
bad. I will rename them in another commit.
Access is protected only for setting/creating/destroying the display
handle. set_caps() for example is not protected because it cannot be
called before changing state to READY, at which point there will be
a display handle available and which cannot change by any thread at
that point
This is because:
* GST_ELEMENT_WARNING/ERROR do lock the OBJECT_LOCK and we deadlock instantly
* In future commits I want to make use of GstBaseSink functions that also
lock the OBJECT_LOCK inside this code
* own_surface is not needed anymore
* gst_wl_window_from_surface is not used externally anymore
* many initializations to 0 are not needed (GObject does them)
This means that the given surface in set_window_handle can now be
the window's top-level surface on top of which waylandsink creates
its own subsurface for rendering the video.
This has many advantages:
* We can maintain aspect ratio by overlaying the subsurface in
the center of the given area and fill the parent surface's area
black in case we need to draw borders (instead of adding another
subsurface inside the subsurface given from the application,
so, less subsurfaces)
* We can more easily support toolkits without subsurfaces (see gtk)
* We can get properly use gst_video_overlay_set_render_rectangle
as our api to set the video area size from the application and
therefore remove gst_wayland_video_set_surface_size.
This drops the ugly GstWaylandWindowHandle structure and is much
more elegant because we can now request the display separately
from the window handle. Therefore the window handle can be requested
in render(), i.e. when it is really needed and we can still open
the correct display for getting caps and creating the pool earlier.
This change also separates setting the wl_surface from setting its size.
Applications should do that by calling two functions in sequence:
gst_video_overlay_set_window_handle (overlay, surface);
gst_wayland_video_set_surface_size (overlay, w, h);
This is the only way to get the negotiation working with the dynamic
detection of formats from the display, because the pipeline needs
to know the supported formats in the READY state and the supported
formats can only be known if we open the display.
Unfortunately,in wayland we cannot have a separate connection to
the display from the rest of the application, so we need to ask for a
window handle when going to READY in order to get the display from it.
And since it's too early to create a top level window from the state
change to READY, create it in render() when there is no other window.
This also changes set_window_handle() to not support window handle
changes in PAUSED/PLAYING (because it's complex to handle and useless
in practice) and make sure that there is always a valid display pointer
around in the READY state.
This fixes weird freezes because of frame_redraw_callback() not being
called from the main thread when it should with weston's toy toolkit.
It's also safer to know that frame_redraw_callback() will always be
called from our display thread... Otherwise it could be called after
the sink has been destroyed for example.
We are not supposed to redraw until we receive a frame callback and this
is especially useful to avoid allocating too many buffers while the
window is not visible, because the compositor may not call wl_buffer.release
until the window becomes visible (ok, this is a wayland bug, but...).
This is achieved by adding an extra reference on the buffers, which does
not allow them to return to the pool. When they are released, this reference
is dropped.
The rest complexity of this patch (hash table, mutex, flag, explicit release calls)
merely exists to allow a safe, guaranteed and deadlock-free destruction sequence.
See the added comment on gstwaylandsink.c for details.
start() makes sure that the minimum ammount of buffers requested is allocated.
stop() makes sure that buffers are actually destroyed and prevents
filling the file system when resizing the surface a lot, because the
wayland-shm-* files will stay on the file system as long as the wl_buffers
created out of them are alive.
This is the initial implementation, without the GstVideoOverlay.expose()
method. It only implements using an external (sub)surface and resizing
it with GstWaylandVideo.
The reference to the sink is not really needed anyway in waylandpool,
what matters basically is that the display is active as long as the
pool is active, so we really want to reference the display object
instead of the sink.
* make use of GstBufferPool::start/stop functions to allocate/deallocate memory
* get rid of struct shm_pool and do all operations cleanly inside WaylandBufferPool
* store a GstVideoInfo during configuration instead of the width & height
and use the stride from the video info instead of hardcoding its value
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 can happen if the playlists have moved due to the variant playlist
now being redirected to another target. This currently only works as long
as the referenced playlists don't change in relation to the variant
playlist, and the new location is purely due to a new path triggered by a
new redirection target of the variant playlist, or a new redirection
target of the playlist itself.
https://bugzilla.gnome.org/show_bug.cgi?id=731164
We add a new signal, get-rollover-counter, to the SRTP encoder. Given a
ssrc the signal will return the currently internal SRTP rollover counter
for the given stream.
For the SRTP decoder we have a new SRTP caps parameter "roc" that needs
to be set when a new SRTP stream is created for a given SSRC.
https://bugzilla.gnome.org/show_bug.cgi?id=726861
Expose one more libcurl option: CURLOPT_SSH_HOST_PUBLIC_KEY_MD5.
This allows authenticating the server by the MD5 fingerprint of
the server's public key.
https://bugzilla.gnome.org/show_bug.cgi?id=723167
The parsing function already frees the old value (if any), avoid a double
free by not freeing it before calling the function without setting the
pointer to NULL
Coverity ID: 1212178
The _parse_url function already frees the previous pointer, avoid
freeing it before without setting to null or we have a double free.
Coverity ID: 1212181
Coverity ID: 1212180
Coverity ID: 1212179
Refactor mssdemux to remove uridownloader to use an internal
source element which reduces startup latency and provides smaller
buffers for better buffering management downstream
data does not have to be freed at all here, it's a pointer to
an arbitrary position inside the current line. Also don't reuse
the data variable for anything else, that will cause crashes
in playlists that have the I-frame playlist URI followed by
other attributes.
CID 1212127
Set up a message handling function to be able to catch errors
from the source element and signal the cond to allow the download
loop to retry the download.
Instead, use a source element linked to a ghostpad to provide
smaller buffers and more granular control for downstream
buffering elements while also reducing startup latency
Only the first buffer of a fragment has its timestamp set, so only
update the segment.position when pushing those buffers to avoid
having GST_CLOCK_TIME_NONE set to the position
https://bugzilla.gnome.org/show_bug.cgi?id=729364
Otherwise we will never recover from previous errors, and especially
will never start again after a flushing seek if downstream returned
GST_FLOW_FLUSHING to us.
hlsdemux can't rely on the source to push flushes on a seek on ready
as that might not make sense. So always resort to flushing the
internal proxy pads by pushing flush events from the source's src pad.
Also as the seeking is not required anymore, only seek if there is
really a byte range to be used. And store a ref to the source's
src pad to avoid doing get_static_pad for every fragment.
In decryption scenario, a buffer is always stored to be sent later
to wait for more data or EOS to be able to strip the final bytes
if requested. In case an error hapenned this buffer can be ignored
and not pushed downstream.
Handle some more error cases:
1) When the source element fails to go to ready
2) When decryption fails
3) When there is no source to handle a specific URI
4) When the URI is invalid
Set up a message handling function to catch errors from the internal
source and store the last return code to identify error situations
when returning from a fragment download.
Also moves the duration increase to after the download when we
know if it was successful or not
When using the internal source, hlsdemux doesn't know the caps of
the input before adding the pad, so remove the arguments that would
use that as it is always NULL.
And use an specific flag to signal when a pad switch is required.
Using the discont flag is a bad idea now because when a fragment
download fails it will lead to exposing a pad group without any
data, causing decodebin to abort.
When receving EOS from the internal src, increase the current positon
by the fragment duration to allow correct restoring of download position
if the bitrate changes
Use the same properties as uridownloader to keep connections alive
between consecutive fragments downloads.
1) set keep-alive property to true
2) keep the element in READY instead of in NULL
Measure the download bitrate to be able to select
the best playlist.
As the buffers are directly pushed downstream and it
might block. The time is only measured from the download
until the pad push and it is started again after the push
returns.
Now the decryption is done buffer by buffer instead of on the
whole fragment at once. As it expects multiples of 16 bytes a
GstAdapter was added to properly chunk the buffers.
Also the last buffer must be resized depending on the value of the
last byte of the fragment, so hlsdemux always keeps a pending buffer
as it doesn't know if it is the last one yet
The GstElement is directly linked into a ghost pad and
its buffers are pushed as received downstream. This way the
buffers are small enough and not a whole fragment that usually
causes extra latency and makes buffering harder
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.
Previously if the proxy server hostname was the empty string
curlhttpsink would never even set the libcurl option. For libcurl
however, having a proxy server hostname be the empty string means that
proxying should be disabled even if environment variables might be set.
Now with the restriction lifted, doing this is allowed.
https://bugzilla.gnome.org/show_bug.cgi?id=728960
rtcp_buffer_get_ssrc is called even with RTP buffers. this means we
might end up with an exception and not find any valid RTCP packet type
and thus hit GST_RTCP_TYPE_INVALID. we now take care of this.
https://bugzilla.gnome.org/show_bug.cgi?id=727512
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.
But only add this for non-live playlists. For live playlists we already
have another thread that is periodically updating playlists.
Reason for this is that sometimes downloading a fragment can fail because
the URIs have changed or expired since last time.
Sequence numbers in different playlists are not guaranteed to be the same for the
same position, e.g. fragments could have different durations in different playlists.
In theory we should do exactly the same for live playlists, but unfortunately we can't
because doing this kind of seeking requires the complete playlist since we started
playback. For live playlists the server is however dropping fragments in the beginning
over time and we have no absolute time references.
The tag was dereferenced earier. From the libschroedinger code,
it's not obvious to see whether tag and frame would be NULL at
the same time. I think is likely that both will be non NULL
here, but that's not certain. Additional tests may be needed
to avoid dereferencing tag and/or frame, but what to do if
only one is NULL isn't obvious, as the _get_tag function does
transfer ownership so isn't undoable.
Coverity 1139850
When we'd see an unknown stream type, then a SDDS stream.
Then we'd get to the end of the switch with a NULL temp stream
pointer, and dereference it.
Coverity 1139708
Recent refactoring causes this code to be called with either a NULL
fragment, or a non NULL fragment. In the former case, we don't have
a buffer. In the latter case, the original code dealing with DISCONT
assumed the buffer was valid. Testing for a NULL buffer here thus
does not seem to change the intent, and fixes:
Coverity 1195147
Turns out there was the same issue as with subtitles.
There is space for a single audio stream, but up to 255
may be used based on a uint8_t value in a struct, which may
or may not be read from the (untrusted) data.
A comment in ifo_types.h says this value is either 0 or 1, so
we can ensure this here without drawbacks.
Coverity 1139585
There is space for a single subtitle stream, but up to 255
may be used based on a uint8_t value in a struct, which may
or may not be read from the (untrusted) data.
A comment in ifo_types.h says this value is either 0 or 1, so
we can ensure this here without drawbacks.
Coverity 1139586
There is a small chance that we might end up in the done step without
having any output available.
Furthermore, when going through not_ready, we need to ensure gst_buffer_unmap
has a properly initialized GstMapInfo.
CID #1139923
CID #1139924
CID #1139919
CID #1139920
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
We create our textures (in Desktop GL) with GL_TEXTURE_RECTANGLE,
vaapi attempts to bind our texture to GL_TEXTURE_2D which throws a
GL_INVALID_OPERATION error and as thus, no video.
Also, by moving exclusively to GL_TEXTURE_2D and the npot extension
we also remove a difference between the Desktop GL and GLES2 code.
https://bugzilla.gnome.org/show_bug.cgi?id=712287
Fix bug #310775
gst-launch audiotestsrc ! libvisual_gl_projectM ! glimagesink is working
but for now you cannot append any other opengl filters between
libvisual_gl_projectM and glimagesink because our FBO is turned OFF.
It would require that libvisual allows to split rendering between
pass1,2,3... and final rendering. In order to unbind our FBO before
the passN, and then rebind it just before the final libvisual rendering.
hlsdemux causes a null pointer dereference if the media playlist
does not contain any media files. The gst_m3u8_client_get_duration
function assumes that demux->client->current->files is valid when
computing duration.
gst_m3u8_client_update needed to be modified to check for the
case of downloading an M3U8 file that doesn't contain any media
files, and returning an error to gsthlsdemux.c
This bug can be reproduced by creating a master m3u8 file that
contains one media playlist that points back to the master m3u8
file. For example create a file called bug725134.m3u8:
#EXTM3U
#EXT-X-VERSION:4
#EXT-X-STREAM-INF:PROGRAM-ID=1, BANDWIDTH=1251135, CODECS="avc1.42001f mp4a.40.2", RESOLUTION=640x352
bug725134.m3u8
https://bugzilla.gnome.org/show_bug.cgi?id=725134
hlsdemux does not check for the '"' character in #EXT-X-STREAM-INF
attributes. The CODECS parameter is an example of an attribute
that might use the '"' symbol and might contain a ',' character
inside this quoted string.
For example: CODECS="avc1.77.30, mp4a.40.2"
hlsdemux does not correctly parse the RESOLUTION attribute, it
assumes that an '=' character is used to delineate the width
and height values, but the HLS RFC states that a 'x' character
must be used as the delimiter between width and height.
https://bugzilla.gnome.org/show_bug.cgi?id=725140
...instead of adding them from the start of playlist every time. This
among other things fixes timestamps for live streams, where the playlist
is some kind of ringbuffer of fragments and thus adding from the beginning
of the playlist will miss the past fragments.
https://bugzilla.gnome.org/show_bug.cgi?id=724983
We now download fragments as fast as possible and push them downstream
while another thread is just responsible for updating live playlists
every now and then.
This simplifies the code a lot and together with the new buffering
mode for adaptive streams in multiqueue makes streams start much faster.
Also simplify threading a bit and hopefully make the GstTask usage safer.
Incorrect time scaling in gst_dash_demux_wait_for_fragment_to_be_available()
means that media segments are fetched before their availablity time. This
patch fixes this.
https://bugzilla.gnome.org/show_bug.cgi?id=724875
demux->last_manifest_update is not initialised at startup, with the
effect that live manifests are reloaded immediately after the download
loop begins. This patch fixes this.
https://bugzilla.gnome.org/show_bug.cgi?id=724790
And only afterwards wait until a fragment was played. Otherwise we're keeping
our cache most of the time at "fragments-cache" fragments minus one.
Also allow setting "fragments-cache" to 1 now to start playback even faster.
