When running GStreamer from uninstalled sources, the location of the haar
cascade files will be local. Check if running in uninstalled and set the
file paths accordingly.
Move handling of a GstSample in a separate function, and unref the
sample after calling it. libass copies the font data so we don't need to
keep it around.
https://bugzilla.gnome.org/show_bug.cgi?id=755759
Ignore the normal gap threshold for laggy streams and
immediately catch all streams up to the end of the segment
when processing gap updates for a segment during a
still frame sequence.
https://bugzilla.gnome.org/show_bug.cgi?id=755680
Create src pads for Representations that contain timed-text subtitles,
both when the subtitles are encapsulated in ISO BMFF (i.e., the
Representation has mimeType "application/mp4") and when they are
unencapsulated (i.e., the Representation has mimeType
"application/ttml+xml").
https://bugzilla.gnome.org/show_bug.cgi?id=747774
The same has to be done for AdaptationSet and SegmentList nodes still.
Also this does not correctly implement the semantics: by default Period (and
other nodes) should only be loaded when needed, not in the very beginning. We
need to implement lazy loading for them, which means adjusting
gst_mpd_client_setup_media_presentation().
https://bugzilla.gnome.org/show_bug.cgi?id=752230
gst_uri_join_strings() will return the second parameter if it is an absolute
URI. No need to do a (wrong) check if the URI is absolute or not beforehand.
https://bugzilla.gnome.org/show_bug.cgi?id=755134
In case the format changed fast and the pending format is different
than the currently set but the currently set is equal to the pending
one we could end up having mismatch between the finally set format
and the data stream format.
https://bugzilla.gnome.org/show_bug.cgi?id=755542
The spec defines these as signed in 5.3.9.6.1.
Since we don't support this behavior, warn and default to 0
(non repeating), which is the spec's default when the value
is not present.
https://bugzilla.gnome.org/show_bug.cgi?id=752480
Even if it doesn't actually advance the subfragment in the default way
for streams that have subfragments, it can help the base class to return
EOS when there is no more fragments instead of signaling it that it should
continue downloading.
https://bugzilla.gnome.org/show_bug.cgi?id=755042
This reverts commit 626a8f0a74.
This allows us to get the plain presentation offset and the period start time
separately. We have to adjust the timestamp by the presentation offset, but
the period start time should only adjust the stream time and running time.
https://bugzilla.gnome.org/show_bug.cgi?id=752409
This reverts commit e671ad25a9.
The timestamps should restart at 0 again for each period, but we have to
adjust the segment to map those timestamps to the actual stream time and
running time of that period.
Otherwise we would have timestamps that conflict with the ones from the tfdt
inside the MP4 container, which are restarting at 0 for each period.
https://bugzilla.gnome.org/show_bug.cgi?id=752409
In dash isombff profile the fragment is split into subframents where
bitrate switching is possible. Also take that into consideration
when checking if a stream has next fragments.
This GstStreamPeriod start value is expressed in nanoseconds,
and the glib time addition function expects microseconds.
There seems to have been a confusion with GstPeriodNode's start
field, which is expressed in milliseconds.
Additionally, add a warning if the timestamp modification did
not succeed, and NULL was returned.
The cvSmooth cvNot functions and do not have the correct input parameters.
Furthermore, cvSmooth function is not necessary for edge detection,
because the Canny function makes the step of smoothing the image.
And cvNot function is useless because there aren't changes if this
function is eliminated.
https://bugzilla.gnome.org/show_bug.cgi?id=754148
The cascade classifier changes its structure on new version of OpenCV 2.4.11.
It is need to migrate to C++ to utilize the new load method of OpenCV which
allows to load the old and new classifiers.
https://bugzilla.gnome.org/show_bug.cgi?id=753994
For PROP_PROFILE case that exist inside gst_face_blur_set_property
function loads the new XML file in the CvHaarClassifierCascade property
without first checking that it is released because maybe there is an XML
file previously loaded.
https://bugzilla.gnome.org/show_bug.cgi?id=753994
Changes inside the gst_face_blur_load_profile function, the number of
input parameters and in lines where it is used due to it cannot be used
generically.
https://bugzilla.gnome.org/show_bug.cgi?id=753994
Change the gstfaceblur.c file to cpp and add it into Makefile.
It is necessary to migrate the faceblur plugin to C++,
in order to load new and old classifiers, to make faceblur work
with newer versions of Opencv.
https://bugzilla.gnome.org/show_bug.cgi?id=753994
Fix some very dubious code. The class methods should always
be set, and the instance-specific check should then be done
inside the method. For data_received that's there already, for
finish_fragment we need to add it.
https://bugzilla.gnome.org/show_bug.cgi?id=753937
Some live streams (eg youtube) don't remove fragments in order to allow
seeking back in time (live + vod).
When gst_m3u8_client_has_next_fragment is called, we are getting wrong fragment
because current_file points in first file of the fragments list resulting in
watching the stream from the beginning again.
This patch sets current_file to nth fragment for live streams, then on
gst_m3u8_client_has_next_fragment will keep up with the live stream.
https://bugzilla.gnome.org/show_bug.cgi?id=753344
This allow properly copying selected meta, like the composition
overlay. Note that output buffer need to be readable, but GlUpload
keeps a ref. For now, simply drop GlUpload ref after perform,
leaving that ref has no purpose. The method shall be removed
in the future.
https://bugzilla.gnome.org/show_bug.cgi?id=754047
Use base class default method instead of only copying flags and
timestamp. This way, selected meta's like compostion overlay will
be passed downstream as expected.
https://bugzilla.gnome.org/show_bug.cgi?id=754047
During allocation query, when this element is not passthrough, it must
relay the overlay compostion meta and it's parameters. Fortunatly, base
transform can do this for us.
https://bugzilla.gnome.org/show_bug.cgi?id=753850
Previous patch did not handle the case where an encoding (e.g. UTF-8) is
specified in the <xml ?> element. Added an extra test for with and without
encoding.
https://bugzilla.gnome.org/show_bug.cgi?id=753813
When running on an STB, the function
gst_mpdparser_get_xml_node_as_string causes a segmentation fault. This
code works correctly on a Linux desktop.
Looking at the libxml documentation, the xmlNodeDump is deprecated.
Replacing the use of xmlNodeDump with xmlNodeDumpOutput fixes the
segfault on the STB and removes the use of the deprecated function.
Some distributions store OpenCV files in /usr/share/opencv and some others
(and default when building from source) install them in
/usr/share/OpenCV. Support both to find cascade files.
https://bugzilla.gnome.org/show_bug.cgi?id=753651
It is faster than doing a query that propagates downstream and
should be enough
Elements: faac, gsmenc, opusenc, sbcenc, voamrwbenc, adpcmenc, sirenenc
When seeking to the last second of a mpd it would reject the seek
because the comparison was < instead of <=
This fails the important use case of seeking to the end of a file
to play it back in reverse from the end
We need to keep the active buffer (the one we have retreive a
texture id from) otherwise it's racy and upstream may upload
new content before we have rendered or during later redisplay.
The urn:mpeg:dash:utc:http-head:2014 method of time synchronisation
uses an HTTP HEAD request to a specified URL and then parses the
Date: HTTP response header.
This commit adds support to dashdemux for this method of time
synchronisation by making a HEAD request and then parsing the Date:
response.
This commit adds support to gstfragment to return the HTTP headers
and to uridownloader to support HEAD requests. To avoid creating a
new API, the RANGE get function is re-used (abused?) with start=-1
and end=-1 to indicate a HEAD request.
https://bugzilla.gnome.org/show_bug.cgi?id=752413
This commit addresses the following items from the code review:
use a portable way to define NTP_TO_UNIX_EPOCH,
fix memory leak on error, and
add documentation to UTCTiming parse functions
Using LL is not portable, so the G_GUINT64_CONSTANT needs to be instead.
If an error occurs during DNS resolution, the GError was not being
released, causing a memory leak.
https://bugzilla.gnome.org/show_bug.cgi?id=752413
Unless the DASH client can compensate for the difference between its
clock and the clock used by the server, the client might request
fragments that either not yet on the server or fragments that have
already been expired from the server. This is an issue because these
requests can propagate all the way back to the origin
ISO/IEC 23009-1:2014/Amd 1 [PDAM1] defines a new UTCTiming element to allow
a DASH client to track the clock used by the server generating the
DASH stream. Multiple UTCTiming elements might be present, to indicate
support for multiple methods of UTC time gathering. Each element can
contain a white space separated list of URLs that can be contacted
to discover the UTC time from the server's perspective.
This commit provides parsing of UTCTiming elements, unit tests of this
parsing and a function to poll a time server. This function
supports the following methods:
urn:mpeg:dash:utc:ntp:2014
urn:mpeg:dash:utc:http-xsdate:2014
urn:mpeg:dash:utc:http-iso:2014
urn:mpeg:dash:utc:http-ntp:2014
The manifest update task is used to poll the clock time server,
to save having to create a new thread.
When choosing the starting fragment number and when waiting for a
fragment to become available, the difference between the server's idea
of UTC and the client's idea of UTC is taken into account. For example,
if the server's time is behind the client's idea of UTC, we wait for
longer before requesting a fragment
[PDAM1]: http://www.iso.org/iso/home/store/catalogue_tc/catalogue_detail.htm?csnumber=66068
dashdemux: support NTP time servers in UTCTiming elements
Use the gst_ntp_clock to support the use of an NTP server.
https://bugzilla.gnome.org/show_bug.cgi?id=752413
This reverts commit ff11a1a8a0.
It can't be assumed that all buffers in a buffer list have the same SSRC or
are RTP or RTCP only. It has to be checked for every single buffer, and one
basically has to do the processing that is done by the default chain_list
implementation.
The current code was ignoring the par/dar aspect when transforming
from window coordinates to stream coordinates resulting in incorrect
coordinates being sent upstream in the navigation events.
The payloader didn't copy anything so far, the depayloader copied every
possible meta. Let's make it consistent and just copy all metas without tags or
with only the audio tag.
https://bugzilla.gnome.org/show_bug.cgi?id=751774
Checking the vector is not empty and checking the vector size is greater
than zero are the same thing, this is a redundancy in the code. Only
checking the vector is not empty is sufficient, therefore removing the
other check.
https://bugzilla.gnome.org/show_bug.cgi?id=744763
Add a pivot vector for setting the origin of rotations and scales.
With the pivot point the rotation and scale operations can have
different origins. This adds the ability to rotate around different points.
Currently the default (0, 0) pivot point is possible,
a rotation around the center, and zooming into and out of the center.
With an pivot point this is optional.
I defined the following image coordinates for the pivot point:
(-1,1) ------------------------- (1,1)
| |
| |
| |
| (0,0) |
| |
| |
| |
(-1,-1) ------------------------- (1,-1)
Example:
Rotate the video at the bottom left corner
gst-launch-1.0 videotestsrc \
! gltransformation \
scale-x=0.5 \
scale-y=0.5 \
rotation-z=25.0 \
pivot-x=-1.0 \
pivot-y=-1.0 \
! glimagesink
The pivot-z option defines the pivot point in 3D space.
This only affects rotation, since we have no Z data to scale.
With this option a video can be rotated around a point in 3D space.
Example:
Rotate around point behind the video:
gst-launch-1.0 videotestsrc \
! gltransformation \
rotation-x=10.0 \
pivot-z=-4.0 \
! glimagesink
Since the profile gchar depends on DEFAULT_FACE_PROFILE, it should never be
NULL. Furthermore CascadeClassifier accepts any input, even
an empty one, but if the profile fails to load it returns an empty cascade.
Check for this instead, and inform the user if there was an Error.
The gst_dash_demux_stream_update_fragment_info function could call
gst_dash_demux_stream_update_headers_info function twice. The
gst_dash_demux_stream_update_headers_info function will set header_uri and
index_uri to some newly allocated strings. The values set by the first call of
gst_dash_demux_stream_update_headers_info will leak when the function is
called for a second time.
The solution is to call gst_adaptive_demux_stream_fragment_clear before the
second call of gst_dash_demux_stream_update_headers_info
https://bugzilla.gnome.org/show_bug.cgi?id=753188
Check if profile is NULL before dereferencing it with new. Also, new will
never return NULL; if allocation fails, a std::bad_alloc exception will be
thrown instead. Remove check for a NULL return.
CID #1315258
The ref_object and object parameters were the wrong way around.
For the typical use case where an application is setting a
GstControlBinding on the returned ghost pad:
1. our control binding would be removed when the new one was set
2. sync_values calls were not being forwarded from the internal
pad to the ghost pad.
If an application attempts to perform other control binding
operations (get_* family of functions) on the internal pad, they
will also be forwarded to the ghost pad where a possible
GstControlBinding will provide the necessary values.
Only copy the values from the parent if the current node doesn't
have that value, they were being copied from the parent and
then overwriten by the child node, leaking the parent's copy
With the switch of gstopencv.c to C++, all OpenCV elements are built with
g++. The template variable clashes with C++'s feature of the same name.
Rename template to templ to avoid any clash.
The cascade classifier changes its structure on new version of OpenCV.
The need to migrate to C++ to utilize the new load method of OpenCV which
allows to load the new classifiers.
https://bugzilla.gnome.org/show_bug.cgi?id=748377
This is used to proxy GstControlBinding to the pad on the
parent object. This avoid having to sync the values in the proxy pad,
this is too early if you have a queue between the pad and the actual
aggregation operation.
https://bugzilla.gnome.org/show_bug.cgi?id=734060
When the sink does not know the window size (e.g not created yet)
it will not add any param to the the composition meta. This is no
reason not to forward this meta API. Fixes issue where it could not
attach until we resize the window.
https://bugzilla.gnome.org/show_bug.cgi?id=745107
The coordinate are relative to the texture dimension and not
the window dimension now. There is no need to pass the window
dimension or to update the overlay if the dimension changes.
https://bugzilla.gnome.org/show_bug.cgi?id=745107
Adds an GST_VIDEO_OVERLAY_COMPOSITION_META_API_TYPE query to glupload
and glimagesink. Detects the query from the downstream elements, so
it is executed only when downstream supports the overlay API.
This makes pipelines with textoverlay ! glupload ! gldownload ! xvimagesink possible.
Uses allocation meta struct for passing the window size upstream.
https://bugzilla.gnome.org/show_bug.cgi?id=745107
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
If a ContentProtection element is present in an AdaptationSet element,
send Protection events on the source pad, so that qtdemux can use this
information to correctly generate its source caps for DASH CENC
encrypted streams.
This allows qtdemux to support CENC encrypted DASH streams where the
content protection specific information is carried in the MPD file
rather than in pssh boxes in the initialisation segments.
This commit adds a new function to the adaptivedemux base class to allow
a GstEvent to be queued for a stream. The queue of events are sent the
next time a buffer is pushed for that stream.
https://bugzilla.gnome.org/show_bug.cgi?id=705991
They require to get_proc_address some functions through the
platform specific {glX,egl}GetProcAddress rather than the default
GL library symbol lookup.
Document that "widget" property must be accessed from the
main thread (where GTK is running). This is the same for
state transition on these elements. It is very natural to
do so un GTK applications.
This is a "pseudo" base class. Basically it's a shared instance
and class structure and a shared set of function between the
two widget. It cannot have it's own type like normal base class
since the one instance will implement GtkGLArea while the other
implements GtkDrawingAreay. To workaround this, the parent instance
and class is a union of both.
https://bugzilla.gnome.org/show_bug.cgi?id=752441
I notice that if you stop the pipeline during a renegotiation
the upload may be NULL while an allocation query is being run.
In that scenario, returning FALSE to the allocation query is the
best thing.
Move back the default property at the same place they are in the
other sink. This helps when using a diff viewer to synchronized
this unfortunate copy paste.
https://bugzilla.gnome.org/show_bug.cgi?id=751104
In GTK dispose can be called before the last ref is reached. This
happens when you close the container window. The dispose will be
explicitly called, and destroyed notify will be fired. This patch
fixes this race by properly tracking the widget state.
In the sink, we now set the widget pointer to NULL, so the widget
will properly get created again if you set your pipeline to NULL
state after the widget was destroy, and set it back to PLAYING.
https://bugzilla.gnome.org/show_bug.cgi?id=751104
Very much in the same spirit as the Gtk GL sink
Two things are provided
1. A QQuickItem subclass that renders out RGBA filled GstGLMemory
buffers that is instantiated from qml.
2. A sink element that will push buffers into (1)
To use
1. Declare the GstGLVideoItem in qml with an appropriate
objectName property set.
2. Get the aforementioned GstGLVideoItem from qml using something like
QQmlApplicationEngine engine;
engine.load(QUrl(QStringLiteral("qrc:/main.qml")));
QObject *rootObject = engine.rootObjects().first();
QQuickItem *videoItem = rootObject->findChild<QQuickItem *> ("videoItem");
3. Set the videoItem on the sink
https://bugzilla.gnome.org/show_bug.cgi?id=752185
Checking for a parent is not enough, it must have a toplevel one.
If widget has no toplevel parent then add it in a GtkWindow, that
make it usable from gst-launch-1.0.
https://bugzilla.gnome.org/show_bug.cgi?id=751104
Moved gst_mpd_client_get_next_segment_availability_end_time and
gst_mpd_client_add_time_difference functions to be grouped with
functions from the same category.
https://bugzilla.gnome.org/show_bug.cgi?id=752027
Corrected the initialisation of mimeType in
gst_mpdparser_get_list_and_nb_of_audio_language: the variable is used
in a loop, so it must be set to NULL at the beginning of each iteration.
https://bugzilla.gnome.org/show_bug.cgi?id=751911
Before returning the next fragment duration value, the
gst_mpd_client_get_next_fragment_duration function tries to validate it.
But the condition was incorrect.
https://bugzilla.gnome.org/show_bug.cgi?id=751539
We're interested in the offset between the period start timestamp and the
actual media timestamp so that we can properly correct for it. The absolute
presentation offset to timestamp 0 is useless as the only thing we really
care about is the offset between the current fragment timestamp and the
media timestamp.
Otherwise we will look for segments after the period usually. The seek
timestamp is relative to the start of the first period and we have to
select a segment relative to the current period's start.
We didn't do this for fragments that are generated on demand from a template,
only for the other cases when they were all generated upfront. This caused
fragment timestamps to start from 0 again for each new period.
If not set, the timeShiftBufferDepth has a default value of -1.
The standard says that this should be interpreted as infinite.
The gst_mpd_client_check_time_position function incorrectly compares
timeShiftBufferDepth with 0 instead of -1 to determine if it was set.
https://bugzilla.gnome.org/show_bug.cgi?id=751500
The last parameter of gst_mpd_client_add_media_segment function is a
duration. But when called from gst_mpd_client_setup_representation, the
last argument was wrongly set to PeriodEnd
https://bugzilla.gnome.org/show_bug.cgi?id=751449
The period start information, calculated in gst_mpd_client_setup_media_presentation
function is stored in stream_period->start. The information read from
xml file and stored in stream_period->period->start is not changed.
If the xml file does not contain the period start information,
stream_period->period->start will be -1.
The function gst_mpd_client_get_next_segment_availability_end_time wants to
use period start time, but incorrectly uses stream_period->period->start
(value from xml file, which could be -1) instead of stream_period->start
(computed value)
https://bugzilla.gnome.org/show_bug.cgi?id=751465
According to ISO/IEC 23009-1:2014(E), chapter 5.3.2.1
"The Period extends until the PeriodStart of the next Period, or until
the end of the Media Presentation in the case of the last Period."
This means that a configured value for optional attribute period duration
should be ignored if the next period contains a start attribute or it is
the last period and the MPD contains a mediaPresentationDuration attribute.
https://bugzilla.gnome.org/show_bug.cgi?id=750797
Support video with multiview info in the caps, transform
it to mono anaglyph by default, but allow for configuring
other output modes and handoff to the app via
the draw signal.
https://bugzilla.gnome.org/show_bug.cgi?id=611157
Added some warning messages in gst_mpd_client_setup_streaming to help
debug situations when the function will return FALSE.
Renamed a wrongly spelled variable.
https://bugzilla.gnome.org/show_bug.cgi?id=751149
Corrected some comments in gstmpdparser.h file.
Moved gst_mpd_client_get_adaptation_sets function to be grouped with
other functions from AdaptationSet group
https://bugzilla.gnome.org/show_bug.cgi?id=751149
The gst_mpdparser_get_rep_idx_with_max_bandwidth function assumes
representations are ordered by bandwidth and incorrectly returns the
first one when wanting the one with minimum bandwidth.
Corrected gst_mpdparser_get_rep_idx_with_max_bandwidth function to get the
correct representation in case max_bandwidth parameter is 0.
https://bugzilla.gnome.org/show_bug.cgi?id=751153
Getting the current viewport and modifying it relatively will produce an
interesting feedback loop during widget resizing. Over a few frames we
will gradually move the viewport a bit until it converged again, adding
unnecessary additional borders at the top and left.
We now know that pool caching can cause renegotiation issues
when an element in the pipeline change from passthrough to not
passthrough. As it's not needed, don't cache existing pools.
https://bugzilla.gnome.org/show_bug.cgi?id=748344
Added a check for a_node->ns before accessing a_node->ns->href in
gst_mpdparser_get_xml_node_namespace. This could happen if the xml
is missing the default namespace.
https://bugzilla.gnome.org/show_bug.cgi?id=750866
If the presentationTimeOffset attribute of a DASH manifest contains
a value that is larger than 2^32, gstmpdparser incorrectly calculates
the stream's presentation time offset. This is due to two bugs:
1: Using gst_mpdparser_get_xml_prop_unsigned_integer rather than
gst_mpdparser_get_xml_prop_unsigned_integer_64 to parse the
attribute
2: gst_mpd_client_setup_representation multiplying the value by
GST_SECOND and then dividing by timescale
https://bugzilla.gnome.org/show_bug.cgi?id=750804
This patch allow going gst-inspect-1.0 on these elements removing
ugly crash that was previously occurring. The method consist of
making the widget creation as lazy as possible. This way we don't
endup doing gtk_init() before the application. We also ref_sink()
the widget, so we don't crash if the parent widget is discarded,
and cleanly error out with GL if the widget has no parent window,
because calling gtk_widget_realized() can only be done if the widget
has been parented to a window).
This reverts commit 4ca3a22b6b.
The connection-speed=0 is used as a special value in the property
of hlsdemux to mean 'automatic' selection, m3u8.c doesn't need
to know about that as it should be as simple as possible.
So this patch hides this automatic selection documented in hlsdemux
into m3u8 logic and I think the gets harder to understand the code.
It also makes the hlsdemux unit tests work again
https://bugzilla.gnome.org/show_bug.cgi?id=749328
This reverts commit 37011e5198.
This change was actually completely unnecessary, the streams in question are
marked as static and are not considered live anyway.
Otherwise we'll only get half of its bits printed on 32 bit architectures.
For this, promote the %d-style format strings to something that accepts
64 bit integers with G_GINT64_MODIFIER.
Using format strings from an untrusted source without validation is
calling for problems, and at least allows to remotely crash your application.
If not worse.
In live situations, it is not uncommon for the current fragment to end
up out of the (updated) play range (lowest/highest sequence). But the next
fragment to play *is* present in the play range.
When advancing, if we can't find the current GstM3U8MediaFile, don't abort
straight away. Instead, look if a GstM3U8MediaFile with the next sequence value
is present, and if so switch to it.
https://bugzilla.gnome.org/show_bug.cgi?id=750028
Previously when compiling GstGL with both GL and GLES2,
GL_RGBA8 was picked from GL/gl.h. But a clash may happen at
runtime when one is selecting GLES2.
gst_gl_internal_format_rgba allows to check at runtime
if it should use GL_RGBA or GL_RGBA8.
The functions to get the next fragment, next fragment timestamp and to advance
to the next fragment need to work differently when stream->segments is NULL.
Use logic similar to that introduced by commit 2105a310 to perform these
functions.
https://bugzilla.gnome.org/show_bug.cgi?id=749684
Previously the VPS unit was detected and all next packets where copied
into the header buffer assuming only SPS and PPS would follow. This is
not always true, also other types of NAL units follow the VPS unit and
where copied to the header buffer. Now the VPS/SPS/PPS are explicitely
detected and copied in the header buffer.
1. Set the sync point after the (possible) upload has occured
2. Wait in the correct GL context (the draw context)
Note: We don't add the GL sync meta to the input buffer as it's not
writable and a copy would be expensive.
Similar to the change with the same name for glimagesink
1. Set the sync point after the (possible) upload has occured
2. Wait in the correct GL context (the draw context)
Note: We don't add the GL sync meta to the input buffer as it's not
writable and a copy would be expensive.
The property level has a minimum value of 0. But when we set the level as 0,
it gets an assertion error. The function icvPyrSegmentation8uC3R returns false
if level is set as 0, since the minimum level cant be 0 and thus results in error.
Hence changing the minimum value to 1.
https://bugzilla.gnome.org/show_bug.cgi?id=749525
When all fragments have already been downloaded on a live stream
dashdemux would busy loop as the default implementation of
has_next_fragment would return TRUE. Implement it to correctly
signal if adaptivedemux should wait for the manifest update before
trying to get new fragments.
When updating the manifest the timestamps on it might have changed a little
due to rounding and timescale conversions. If the change makes the timestamp
of the current segment to go up it makes dashdemux reposition to the previous
one causing one extra unnecessary download.
So when repositioning add an extra 10 microseconds to cover for that rounding
issues and increase the chance of falling in the same segment.
Additionally, also improve the time used when the client is already after the
last segment. Instead of using the last segment starting timestamp use the
final timestamp to make it reposition to the next one and not to the one that
has already been downloaded.
These functions of directly getting and setting segment indexes
are no longer useful as now we need 2 indexes: repeat and segment
index.
The only operations needed are advance_segment, going back to the
first one or seeking for a timestamp.
Segments are now stored with their repeat counts instead of spanding
them to multiple segments. This caused advancing to the next segment
using a single index to have to iterate over the whole list every time.
This commit addresses this by storing both the segment index as well
as the repeat index and makes advancing to next segment just an
increment of the repeat or the segment index.
Use a single segment to represent it internally to avoid using too
much memory. This has the drawback of issuing a linear search to
find the correct segment to play but this can be fixed by using
binary searches or caching the current position and just looking
for the next one.
https://bugzilla.gnome.org/show_bug.cgi?id=748369
The custom code is wrong as it ignores the templates, which leads to
missing fields in the result. Instead, simply use the default get_caps
implementation which does it correctly (get the template, intersect
with filter and return).
https://bugzilla.gnome.org/show_bug.cgi?id=749237
Without this, we will fixate weird pixel-aspect-ratios like 1/2147483647. But
in the end, all the negotiation code in videoaggregator needs a big cleanup
and videoaggregator needs to get rid of the software-mixer specific things
everywhere.
Upstream might not give us a caps event (dtlssrtpdec) because it might be an
RTP/RTCP mixed stream, but we split the two streams anyway and should report
proper caps downstream if possible.
Fixes "sticky event misordering" warnings with dtlssrtpdec.
And provide home-made fallback for older GLib versions,
so that we can later find these and remove them when
we bump the GLib requirement (which is certainly going
to happen before 2.0).
https://bugzilla.gnome.org/show_bug.cgi?id=748495
It's better to just select some random variant playlist instead of stopping,
chances are that it's still continuing to work and we might just have to
select a different variant again later.
We should only refresh the currently selected variant playlist (if any,
otherwise the main playlist), not the main playlist. And only try to
refresh the main playlist if updating the variant playlist fails.
Some servers (Wowza) use the request of the main playlist to create a
"session", which is then part of the URI of the variant playlist and
also the fragments. Refreshing the main playlist would generate a new
session, and the server rate limits that usually. And after a few retries
the server just kicks us out.
Also as a side effect we now use the same downloader for all playlists, so
that we only have 2 instead of 3 connections to the server. And also
previously we just ignored the downloaded data from the main playlist that
the base class gave to us.
When the segment is very short it might be the case that the
typefinding fails and when finishing the segment hlsdemux would
consider the remaining data (pending_buffer) as an encryption
leftover.
This patch fixes it and makes sure an error is properly posted
if typefind failed by refactoring buffer handling to a function
and using it from the data_received and finish_fragment functions.
We also have to update the current_file GList pointer in the M3U playlist
client, otherwise we are just continuing playback from the current position
instead of seeking.
Variable hands is already checked to contain a value previously at the beginning
of the current block. There is no need to check again. This is logically dead code.
CID 1197693
Caps refcounting was all wrong in this function. Rewrote it and add some
comments to make it clearer.
Fix caps leaks with the
validate.file.glvideomixer.simple.play_15s.synchronized scenario.
https://bugzilla.gnome.org/show_bug.cgi?id=747915
Signed-off-by: Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
If old opencv1-style legacy include directory is available,
this change becomes purely cosmetic (maybe will compile a bit faster).
It becomes an FTBFS fix when opencv1-style include directory is missing
(possibly because opencv package maintainer decided not to pack it).
https://bugzilla.gnome.org/show_bug.cgi?id=747705
Fix a caps leak with the
validate.file.glvideomixer.simple.play_15s.synchronized scenario.
https://bugzilla.gnome.org/show_bug.cgi?id=747915
Signed-off-by: Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
'array_buffers' contain borrowed GstBuffer and so shouldn't have a free
function. 'frames' is the one containing GstGLMixerFrameData and so should use
_free_glmixer_frame_data as free function.
Fix GstGLMixerFrameData leaks with the
validate.file.glvideomixer.simple.play_15s.synchronized scenario.
https://bugzilla.gnome.org/show_bug.cgi?id=747913
Signed-off-by: Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
Fix a simple buffer overflow - 16 bytes isn't enough to hold
the string representation of a gulong on x86_64. I guess the
intent was to generate a 32 bit random key, so let's do that.
Only matters if anyone ever ports the sink to 1.x
https://bugzilla.gnome.org/show_bug.cgi?id=676524
There is a playback error when trying to play a content that
has 'application' mimeType. This commit prevents an exception from
setup text streams.
https://bugzilla.gnome.org/show_bug.cgi?id=747525
As mentionned in release notes : Added new Sps/Pps strategies for real-time
video (replace the old setting variable 'bEnableSpsPpsIdAddition' with
'eSpsPpsIdStrategy')
upstream might send buffer lists instead of buffers and hlssink's
probe won't get called and a new segment won't be created when needed.
This patch fixes it by adding a chain_list function to the sink pad
that will just pass through the whole bufferlist if no segment needs
to be requested at the moment or convert the list into buffers to
check the proper timestamp to request the next key-unit that will
start the segment.
https://bugzilla.gnome.org/show_bug.cgi?id=746906
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.
No need to ref/unref the connection every time we push something on the pool.
However we have to provide non-NULL data to the pool, so let's just give it
some coffee.
This way we will share threads with other DTLS connections if possible, and
don't have to start/stop threads for timeouts if there are many to be handled
in a short period of time.
Also use the system clock and async waiting on it for scheduling the timeouts.
GST_DTLS_USE_GST_LOG is not defined anywhere, so
we'd just log into the default category by accident.
We use the gst logging system unconditionally now,
so might just as well remove this #if #else.
gcc-4.9.2:
gstdtlsagent.c:114:1: error: old-style function definition
gstdtlsconnection.c:253:3: error: ISO C90 forbids mixed declarations and code
gstdtlsconnection.c:291:3: error: ISO C90 forbids mixed declarations and code
gstdtlsconnection.c:391:3: error: ISO C90 forbids mixed declarations and code
gstdtlsconnection.c:434:3: error: ISO C90 forbids mixed declarations and code
gstdtlsconnection.c:773:1: error: 'BIO_s_gst_dtls_connection' was used with no prototype before its definition
gstdtlsconnection.c:773:1: error: old-style function definition
gstdtlsconnection.c:128:32: error: passing 'const char [30]' to parameter of type 'void *'
discards qualifiers [-Werror,-Wincompatible-pointer-types-discards-qualifiers]
SSL_get_ex_new_index (0, "gstdtlsagent connection index", NULL, NULL,
^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
/usr/include/openssl/ssl.h:1981:43: note: passing argument to parameter 'argp' here
int SSL_get_ex_new_index(long argl, void *argp, CRYPTO_EX_new *new_func,
^
gstdtlsconnection.c:822:40: error: arithmetic on a pointer to void is a GNU extension
[-Werror,-Wpointer-arith]
memcpy (out_buffer, priv->bio_buffer + priv->bio_buffer_offset, copy_size);
~~~~~~~~~~~~~~~~ ^
In some upload implementations the out buffer has more than one references,
turning the buffer not writable, so it won't be possible to modify its
meta-data.
This patch moves the meta-data copy before increasing the reference of the out
buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=746173
glupload ! glcolorconvert ! sink
Some properties are manually forwarded. The rest are available using
GstChildProxy.
The two signals are forwarded as well.
It encapsulates a confiurable GL processing element in the
upload/colorconvert/download dance required to transparently process
the majority of GstBuffer's.
GLImage does not use any kind of internal pool. There was some
remaining code and comment stating that it was managing the
pool, and it was in fact setting the active state when doing
to ready state.
* Only create the pool if requested and in propose_allocation
* Cache the pool to avoid reallocation on spurious reconfigure
* Don't try to deactivate the pool (we don't own it)
https://bugzilla.gnome.org/show_bug.cgi?id=745705
If searchIdx() doesn't find the id it returns -1, which breaks
motioncelssvector.at (idx). Check for it and return if not found.
Changing a few other lines for style consistency.
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
LibJPEG uses macroblock of 8x8 sample. In this element we use RGB and
Y444, two 24bit formats that are stored in 32bit pixels. This mean we
have 32x32 bytes macroblocks. For this reason, we need to allocate
our buffer slightly larger. We also need to pass the line pointer in
the right order, otherwise the image endup upside-down.
https://bugzilla.gnome.org/show_bug.cgi?id=745109
Using mkstemp without setting the permission mask is potentially harmful.
POSIX specification of mkstemp() does not say anything about file modes, so we
need to make sure its file mode creation mask is set appropriately before
calling it.
This implements support for GstAllocationParams and memory alignments.
The parameters where simply ignored which could lead to crash on
certain platform when used with libav and no luck.
https://bugzilla.gnome.org/show_bug.cgi?id=744246
+ Split headers from source
+ Remove uneeded AM_CFLAGS, AM_LDFLAGS
+ Always set OBJCFLAGS
Due to the presence of a .m and regardless of the conditional values,
automake will promote the link command to OBJC using OBJCFLAGS. Only
the basic flags (like warnings and optimization) are going to make a
difference though.
This cleanup builds up the makefile with less specific files first
toward more specific file. FLAGS are built with the basic that unused
flags will have empty variable.
i686-apple-darwin11-llvm-gcc-4.2
gstglmixer.h:43: error: redefinition of typedef ‘GstGLMixer’
gstglmixerpad.h:32: error: previous declaration of ‘GstGLMixer’ was here
gstglmixer.h:46: error: redefinition of typedef ‘GstGLMixerFrameData’
gstglmixerpad.h:33: error: previous declaration of ‘GstGLMixerFrameData’ was here
The graphene-1.0 part should not be in the source code. This directory
is part of the cflags include. This is similar to gstreamer-1.0/
directory. This break compilation if the include directory where
graphene is installed is not in your include path.
Bitrate-limit is already available in the baseclass and, even though
the bandwidth-usage name is better, hls and mss already used
bitrate-limit. This patch deprecates the bandwidth-usage and maps
it to the baseclass bitrate-limite.
Move the property from subclasses to adaptivedemux, it allows
selecing the percentage of the measured bitrate to be used when
selecting stream bitrates
Allow the playlist-length to accept '0' as a value, indicating
that no segment should be removed from the playlist. This allows
generating playlists to be used as VOD when complete.
Allows to set a bitrate directly instead of measuring it internally
based on the received chunks. The connection-speed was removed from
mssdemux and hlsdemux as it is now in the base class
By implementing get_live_seek_range.
As shown by :
gst-validate-1.0 playbin \
uri=http://dev-iplatforms.kw.bbc.co.uk/dash/news24-avc3/news24.php
This patch handles live seeking, by setting a live seek range
comprised between now - timeShiftBufferDepth and now.
The inteersting thing with this stream is that one can actually
ask fragments up to availabilityStartTime, but it seems quite clear
in the spec that content is only guaranteed to exist up to
timeShiftBufferDepth.
One can test live seeking this way :
gst-validate-1.0 playbin \
uri=http://dev-iplatforms.kw.bbc.co.uk/dash/news24-avc3/news24.php \
--set-scenario seek_back.scenario
with scenario being:
description, seek=true
seek, playback-time=position+5.0, start="position-600.0",
flags=accurate+flush
This example will play the stream, wait for five seconds, then seek back
to a position 10 minutes earlier.
https://bugzilla.gnome.org/show_bug.cgi?id=744362
Add parsed/framed=true to allow negotiation with some
muxers that required parsed input. Encoders already provide
parsed/framed output so it should say so in caps.
Some variables are not initialized in the constructor. It is highly unlikely
they are used before being set, but it is safer to initialize them.
CID #1197704
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.
Add more power to the chunk_received function (renamed to data_received)
and also to the fragment_finish function.
The data_received function must parse/decrypt the data if necessary and
also push it using the new push_buffer function that is exposed now. The
default implementation gets data from the stream adapter (all available)
and pushes it.
The fragment_finish function must also advance the fragment. The default
implementation only advances the fragment.
This allows the subsegment handling in dashdemux to continuously download
the same file from the server instead of stopping at every subsegment
boundary and starting a new request
gstdashdemux.c:1330:13: error: implicit conversion from enumeration type 'enum _GstAdaptiveDemuxFlowReturn' to different enumeration type
'GstFlowReturn' [-Werror,-Wenum-conversion]
ret = GST_ADAPTIVE_DEMUX_FLOW_SUBSEGMENT_END;
~ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
gmyth seems to be unmaintained upstream, and no one has asked
for this to be ported for a very long time, so let's just
remove it. Neither debian nor Fedora seem to ship libgmyth
any longer, and in any case it's most likely deprecated by
the UPnP support in MythTV.
The segment start time is calculated as the offset into the current segment.
The old condition to detect the end of period (i.e. segment start time >
period start + period duration) failed when the period start was not 0 since
the segment start time does not take the period start time into account.
Fix this detection by only comparing the segment start to the period duration.
https://bugzilla.gnome.org/show_bug.cgi?id=733369
The ISOBMFF profile allows definind subsegments in a segment. At those
subsegment boundaries the client can switch from one representation to
another as they have aligned indexes.
To handle those the 'sidx' index is parsed from the stream and the
entries point to pts/offset of the samples in the stream. Knowing that
the entries are aligned in the different representation allows the client
to switch mid fragment. In this profile a single fragment is used per
representation and the subsegments are contained in this fragment.
To notify the superclass about the subsegment boundary the chunk_received
function returns a special flow return that indicates that. In this case,
the super class will check if a more suitable bitrate is available and will
change to the same subsegment in this new representation.
It also requires special handling of the position in the stream as the
fragment advancing is now done by incrementing the index of the subsegment.
It will only advance to the next fragment once all subsegments have been
downloaded.
https://bugzilla.gnome.org/show_bug.cgi?id=741248
The old code was using gst_caps_normalize() and was generally overly
complex. Simplify by picking sample rate and number of channels from
upstream and the sample format from the allowed caps. If the format caps
is a list of strins, just pick the first one. And if the srcpad isn't
linked yet, use the default format (S16).
https://bugzilla.gnome.org/show_bug.cgi?id=740195
Optimize loop by moving condition outside of it and reuse the
find_next_fragment function to check if there is next instead of
replicating the same loop
Duration queries can be done a few times per second and would cause
the segment list to be traversed for every one. Caching the duration
prevents that.
Variable hands is already checked to contain a value previously at the beginning
of the current block (in line 504). There is no need to check again. This is
logically dead code.
CID 1197693
The duration values in playlists are approximate only, and for
playlist versions 2 and older they are only rounded integer values.
They cannot be used to timestamp buffers. This resulted in playback
gaps and skips because the actual duration of fragments is slightly
different. The solution is to only set the pts of the very first
buffer processed, not for each fragment.
q->bitrate is a guint64, but G_TYPE_INT may read fewer bits
off the stack, and if we pass more then the NULL sentinel
may not be found at the right place, which in turn might
lead to crashes.
https://bugzilla.gnome.org/show_bug.cgi?id=741751
hlsdemux assumes that seeking is not allowed for live streams,
however seek is possible if there are sufficient fragments in the
manifest. For example the BBC have live streams that contain 2 hours
of fragments.
The seek code for both live and on-demand is common code. The
difference between them is that an offset has to be calculated
for the timecode of the first fragment in the live playlist.
When hlsdemux starts to play a live stream, the possible seek range
is between 0 and A seconds. After some time has passed, the beginning of
the stream will no longer be available in the playlist and the seek
range is between B and C seconds.
Seek range:
start 0 ........... A
later B ........... C
This commit adds code to keep a note of the B and C values
and the highest sequence number it has seen. Every time it updates the
media playlist, it walks the list of fragments, seeing if there is a
fragment with sequence number > highest_seen_sequence. If so, the values
of B and C are updated. The value of B is used when timestamping
buffers.
It also makes sure the seek range is never closer than three fragments
from the end of the playlist - see 6.3.3. "Playing the Playlist file"
of the HLS draft.
https://bugzilla.gnome.org/show_bug.cgi?id=725435
For small amounts some data might be mistyped and it would cause
the pipeline to fail. For example if you have AAC inside mpegts,
for small amounts, the AAC samples would cause the typefinder to
think it is AAC and not mpegts.
https://bugzilla.gnome.org/show_bug.cgi?id=736061
If typefind fails, check to see if the buffer is too short for typefind. If this is the case,
prepend the decrypted buffer to the pending buffer and try again the next time around.
https://bugzilla.gnome.org/show_bug.cgi?id=740458
Corrected the final boundary mechanism so that a final boundary is
added to each mail with multipart content that is sent,
not just to the last one.
https://bugzilla.gnome.org/show_bug.cgi?id=741553
This reverts commit 15394aa705.
The latest release (v1.1) does not have pkg-config support
yet, so this plugin won't be built with the latest release.
Cerbero uses the latest release, so this makes cerbero
builds fail, which expect the plugin to be built.
We can re-commit this once there's a release that includes
pkg-config support.
Rework reverse fragment traversing with repetition fields to prevent
NULL pointer deref and avoid never advancing a fragment as the variable
is unsigned and would always be non-negative.
CID #1257627
CID #1257628
Read the "r" attribute from fragments to support fragments nodes
that use repetition to have a shorter Manifest xml.
Instead of doing:
<c d="100" />
<c d="100" />
You can use:
<c d="100" r="2" />
According to the HLS spec the remainder of the line following
the comma on EXTINF tag is not required. This patch removes
the fake title and saves some bytes on the playlist.
https://bugzilla.gnome.org/show_bug.cgi?id=741096
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
Otherwise e.g. videotestsrc ! openh264enc ! ... will drop every second frame
because otherwise the target bitrate can't be reached without loosing too
much quality.
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).
Both Firefox and Chrome uses OPUS as the encoding in their SDP.
Adding this now defacto standard name remove the need for special
case in SDP parsing code.
https://bugzilla.gnome.org/show_bug.cgi?id=737810
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.
Do not try to render a buffer that is already being rendered.
This happens typically during the initial rendering stage as the first
buffer is rendered twice: first by preroll(), then by render().
This commit avoids this assertion failure:
CRITICAL: gst_wayland_compositor_acquire_buffer: assertion
'meta->used_by_compositor == FALSE' failed
https://bugzilla.gnome.org/show_bug.cgi?id=738069
Signed-off-by: Fabien Dessenne <fabien.dessenne@st.com>
Signed-off-by: Benjamin Gaignard <benjamin.gaignard@linaro.org>
If waylandsink is the owner of the display then it is in charge
of catching input events on the surface.
https://bugzilla.gnome.org/show_bug.cgi?id=733682
Signed-off-by: Tifaine Inguere <tifaine.inguere@st.com>
Reviewed-by: Benjamin Gaignard <benjamin.gaignard@linaro.org>
There are two cases covered here:
1) The GstWlDisplay forces the release of the last buffer and the pool
gets destroyed in this context, which means it unregisters all the
other buffers from the GstWlDisplay as well and the display->buffers
hash table gets corrupted because it is iterating.
2) The pool and its buffers get destroyed concurrently from another
thread while GstWlDisplay is finalizing and many things get corrupted.
The main reason behind this is that when the video caps change and the video
subsurface needs to resize and change position, the wl_subsurface.set_position
call needs a commit in its parent in order to take effect. Previously,
the parent was the application's surface, over which there is no control.
Now, the parent is inside the sink, so we can commit it and change size smoothly.
As a side effect, this also allows the sink to draw its black borders on
its own, without the need for the application to do that. And another side
effect is that this can now allow resizing the sink when it is in top-level
mode and have it respect the aspect ratio.
Because we no longer have a custom buffer pool that holds a reference
to the display, there is no way for a cyclic reference to happen like
before, so we no longer need to explicitly call a function from the
display to release the wl_buffers.
However, the general mechanism of registering buffers to the display
and forcibly releasing them when the display is destroyed is still
needed to avoid potential memory leaks. The comment in wlbuffer.c
is updated to reflect the current situation.
This reduces the complexity of having a custom buffer pool, as
we don't really need it. We only need the custom allocation part.
And since the wl_buffer is no longer saved in a GstMeta, we can
create it and add it on the buffers in the sink's render()
function, which removes the reference cycle caused by the pool
holding a reference to the display and also allows more generic
scenarios (the allocator being used in another pool, or buffers
being allocated without a pool [if anything stupid does that]).
This commit also simplifies the propose_allocation() function,
which doesn't really need to do all these complicated checks,
since there is always a correct buffer pool available, created
in set_caps().
The other side effect of this commit is that a new wl_shm_pool
is now created for every GstMemory, which means that we use
as much shm memory as we actually need and no more. Previously,
the created wl_shm_pool would allocate space for 15 buffers, no
matter if they were being used or not.
This also removes the GstWlMeta and adds a wrapper class for wl_buffer
which is saved in the GstBuffer qdata instead of being a GstMeta.
The motivation behind this is mainly to allow attaching wl_buffers on
GstBuffers that have not been allocated inside the GstWaylandBufferPool,
so that if for example an upstream element is sending us a buffer
from a different pool, which however does not need to be copied
to a buffer from our pool because it may be a hardware buffer
(hello dmabuf!), we can create a wl_buffer directly from it and first,
attach it on it so that we don't have to re-create a wl_buffer every
time the same GstBuffer arrives and second, force the whole mechanism
for keeping the buffer out of the pool until there is a wl_buffer::release
on that foreign GstBuffer.
Header will be read each and everytime parse function will be called
which is not necessary since until we have complete data,
we need not parse the header again.
https://bugzilla.gnome.org/show_bug.cgi?id=737984
In gst_hls_demux_get_next_fragment() the next fragment URI gets
stored in next_fragment_uri, but the gst_hls_demux_updates_loop()
can at any time update the playlist, rendering this string invalid.
Therefore, any data (like key, iv, URIs) that is taken from a
GstM3U8Client needs to be copied. In addition, accessing the
internals of a GstM3U8Client requires locking.
https://bugzilla.gnome.org/show_bug.cgi?id=737793
As openh264 has no way to attach any IDs to input frames that we then get on
the output frames, we have to assume that the input has valid PTS. We just
take the frame with the oldest PTS, and if there is no PTS information we take
the one with the oldest DTS.
- 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
If EOS or ERROR happens before the download loop thread has reached its
g_cond_wait() call, then the g_cond_signal doesn't have any effect and
the download loop thread stucks later.
https://bugzilla.gnome.org/show_bug.cgi?id=735663
If EOS or ERROR happens before the download loop thread has reached its
g_cond_wait() call, then the g_cond_signal doesn't have any effect and
the download loop thread stucks later.
https://bugzilla.gnome.org/show_bug.cgi?id=735663
If EOS or ERROR happens before the download loop thread has reached its
g_cond_wait() call, then the g_cond_signal doesn't have any effect and
the download loop thread stucks later.
https://bugzilla.gnome.org/show_bug.cgi?id=735663
The internal pad still keeps its EOS flag and event as it can be assigned
after the flush-start/stop pair is sent. The EOS is assigned from the streaming
thread so this is racy.
To be sure to clear it, it has to be done after setting the source to READY to
be sure that its streaming thread isn't running.
https://bugzilla.gnome.org/show_bug.cgi?id=736012
The internal pad still keeps its EOS flag and event as it can be assigned
after the flush-start/stop pair is sent. The EOS is assigned from the streaming
thread so this is racy.
To be sure to clear it, it has to be done after setting the source to READY to
be sure that its streaming thread isn't running.
https://bugzilla.gnome.org/show_bug.cgi?id=736012
The internal pad still keeps its EOS flag and event as it can be assigned
after the flush-start/stop pair is sent. The EOS is assigned from the streaming
thread so this is racy.
To be sure to clear it, it has to be done after setting the source to READY to
be sure that its streaming thread isn't running.
https://bugzilla.gnome.org/show_bug.cgi?id=736012
packetized mode is being set when framerate is being set
which is not correct. Changing the same by checking the
input segement format. If input segment is in TIME it is
Packetized, and if it is in BYTES it is not.
https://bugzilla.gnome.org/show_bug.cgi?id=736252
Previously we only refetched the playlist if downloading a fragment
has failed once. We should also do that if it failed a second or third time,
chances are that the playlist was updated now and contains new URIs.
face detection will be performed only if image standard deviation is
greater that min-stddev. Default min-stddev is 0 for backward
compatibility. This property will avoid to perform face detection on
images with little changes improving cpu usage and reducing false
positives
https://bugzilla.gnome.org/show_bug.cgi?id=730510
* 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
If the language is not specified in the AdaptationSet, use the ContentComponent
node to get it. We only get it if there is only a single ContentComponent as
it doesn't seem clear on what to do if there are multiple entries
https://bugzilla.gnome.org/show_bug.cgi?id=732237
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.
Instead always use the low bandwith playlist making things go smoother
as the current heuristic is rather set for normal playback, and
currently it does not behave properly.
https://bugzilla.gnome.org/show_bug.cgi?id=734445
When a seek with a negative rate is requested, find the target
segment where gstsegment.stop belongs in and then download from
this segment backwards until the first segment.
This allows proper reverse playback.
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
When flushing, this will prevent dashdemux from trying to download more
fragments or more chunks of the same fragment before stopping.
Also improves the error handling to not transform everything non-ok into
an error.
https://bugzilla.gnome.org/show_bug.cgi?id=734014
templatematch operates on BGR data. In fact, OpenCV's IplImage always
stores color image data in BGR order -- this isn't documented at all in
the OpenCV source code, but there are hints around the web (see for
example
http://www.cs.iit.edu/~agam/cs512/lect-notes/opencv-intro/opencv-intro.html#SECTION00041000000000000000
and http://www.comp.leeds.ac.uk/vision/opencv/iplimage.html ).
gst_templatematch_load_template loads the template (the image to find)
from disk using OpenCV's cvLoadImage, so it is stored in an IplImage in
BGR order. But in gst_templatematch_chain, no OpenCV conversion
functions are used: the imageData pointer of the IplImage for the video
frame (the image to search in) is just set to point to the raw buffer
data. Without this fix, that raw data is in RGB order, so the call to
cvMatchTemplate ends up comparing the template's Blue channel against
the frame's Red channel, producing very poor results.
Previously changing the template property resulted in an exception
thrown from cvMatchTemplate, because "dist_image" (the intermediate
match-certainty-distribution) was the wrong size (because the
template image size had changed).
Locking has also been added to allow changing the properties (e.g. the
pattern to match) while the pipeline is playing.
* gst_element_post_message is moved outside of the lock, because it will
call into arbitrary user code (otherwise, if that user code calls into
gst_templatematch_set_property on this same thread it would deadlock).
* gst_template_match_load_template: If we fail to load the new template
we still unload the previous template, so this element becomes a no-op
in the pipeline. The alternative would be to keep the previous template;
I believe unloading the previous template is a better choice, because it
is consistent with the state this element would be in if it fails to
load the very first template at start-up.
Thanks to Will Manley for the bulk of this work; any errors are probably
mine.
The early return was bypassing the call to gst_pad_push. With no
filter->template (and thus no filter->cvTemplateImage) the rest of this
function is essentially a no-op (except for the call to gst_pad_push).
This (plus the previous commit) allows templatematch to be
enabled/disabled without removing it entirely from the pipeline, by
setting/unsetting the template property.
Delaying the segment event to when caps are decided can cause issues as
the first thing katedec does on its chain function it doing a segment clip.
It will lead to an assertion if the segment format is undefined
https://bugzilla.gnome.org/show_bug.cgi?id=733226
Properly handle the caps event by configuring the kate decoding lib using the
available streamheaders. This makes it possible to decode kate subtitles when
the stream is seeked before katedec gets the initial buffers that are usually
the streamheaders.
https://bugzilla.gnome.org/show_bug.cgi?id=733226
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
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.
First this is handle by base transform, hence this is a no-op, and if it wasn't it
would lead to a buffer copy being leaked, and then an unreffed buffer being
pushed downstream.
https://bugzilla.gnome.org/show_bug.cgi?id=732756
OpenNI2 makes no guarantees of timestamp starting from zero, just that
it will be a millisecond timestamp. Make timestamps start from zero
manually so things work correctly.
https://bugzilla.gnome.org/show_bug.cgi?id=732535
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
Only reset the decryption engine on the first buffer of a fragment,
not again for the second buffer. This fixes corrupting the second
buffer of a fragment.
https://bugzilla.gnome.org/show_bug.cgi?id=731968
gstwaylandsink.c:480:14: error: comparison of constant -1 with expression of
type 'enum wl_shm_format' is always false
[-Werror,-Wtautological-constant-out-of-range-compare]
if (format == -1)
~~~~~~ ^ ~~
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.
Use glib to get a list of system "share" directories, then go through that
list, appending 'sounds/sf2/' to each directory to get a soundfont directory,
and looking for .sf2 files there.
This way fluiddec is able to load sf2 files on W32, because otherwise the
path '/usr/share/sounds/sf2' makes no sense there.
Fixes#724013
nettle is used by newer versions of gnutls, while older versions of gnutls
used libgcrypt. Support both for now as not every distro has nettle yet.
nettle is preferred as it is more efficient to use and much smaller.
This will be incredible slow if the upstream block size is very small. Instead
continue scanning for the header where we previously stopped.
For the standard filesrc block-size this made decoding a file about
3 times faster.
https://bugzilla.gnome.org/show_bug.cgi?id=719890
Merge various changes and fixes from the master mpegdemux
Performance improvement from the way streams are organised,
return flow combining, language tag event generation,
adjustments and fixes in debug output, and things like that.
Previously faces would only be detected if they were at least 30x30 pixels
large and at most 32x32 pixels. We keep the minimum setting (maybe needs
a property as in facedetect) but disable the maximum feature size.
See https://bugzilla.gnome.org/show_bug.cgi?id=722158
This disables the "max feature size" feature. The current configuration
is totally busted: The max feature size is hard-coded to 2 pixels more
than the user-supplied min feature size which pretty much means you need
to guess the size of the person's face to within a few pixels to get the
code to find it.
https://bugzilla.gnome.org/show_bug.cgi?id=722158
Remove the dashdemux seeking function to use the one implemented
in mpdparser as it is more complete. This also makes dashdemux not
crash when seeking on streams that use segment templates.
1275 is the maximum size of a frame, but the encoder may return
up to 3 frames, and we need a few extra bytes for TOC, etc. We
use 4000, which is a bit more, and suggested in the libopus docs.
Download and push from the same task, makes code a lot simpler
to maintain. Also pushing from separate threads avoids deadlocking
when gst_pad_push blocks due to downstream queues being full.
Use a single lock for all streams instead of having separate locks.
This makes maintenance easier and at most points we would need
a single lock before iterating on all streams data. So not much
is gained from individual locks.
Make dash playlists with multiple periods work again by waiting
to switch the periods when all streams have reached the end of
the current period. The stream_loop is responsible for advancing
the period, but the download loops will already start downloading
data for the next period as soon as possible.
Handle multiple languages by using the not-linked return to stop
the download task for that stream. It can be reactivated when
a reconfigure event is received. Stopping the unused streams is
relevant to save network bandwidth
Instead of having a single download task for all streams, this
commit makes each stream have its own download loop, allowing
parallel download of fragments.
always expose all streams instead of only exposing one of each type.
This is more aligned with gstreamer's way of working. Allows the user
to select the stream that it wants to use by linking its pad and leaving
the unused ones as unlinked.
As streams now flow independently, the GstSegment needs to be put
on each stream so they can track the position of each one correctly
instead of being mixed in a single segment
Download and push from the same task, makes code a lot simpler
to maintain. Also pushing from separate threads avoids deadlocking
when gst_pad_push blocks due to downstream queues being full
When a stream gets a not-linked return, it will be marked as so and
won't download any more new fragments until a reconfigure event
is received. This will make mssdemux expose all pads, but only download
fragments for the streams that are actually being used.
Relying on the pads being linked/unlinked isn't enough in this scenario
as there might be an input-selector downstream that is actually discarding
buffers for a given linked pad.
When streams are switching, the old active stream can be blocked because
input-selector will block not-linked streams. In case the mssdemux's
stream loop is blocked pushing a buffer to a full queue downstream it will
never unblock as the queue will not drain (input-selector is blocking).
In this scenario, stream switching will deadlock as input-selector is
waiting for the newly active stream data and the stream_loop that would
push this data is blocked waiting for input-selector.
To solve this issue, whenever an stream is reactivated on a reconfigure
it will enter into the 'catch up mode', in this mode it can push buffers
from its download thread until it reaches the currrent GstSegment's position.
This works because this timestamp will always be behind or equal to the maximum
timestamp pushed for all streams, after pushing data for this timestamp,
the stream will go back to default and be pushed sequentially from the main
streaming thread. By this time, the input-selector should have already
released the thread.
https://bugzilla.gnome.org/show_bug.cgi?id=711849
* ext/srtp/gstsrtp.[ch]: added GST_SRTP_CIPHER_AES_256_ICM to
GstSrtpCipherType and new function cipher_key_size.
* ext/srtp/gstsrtpenc.c: maximum key size is now 46 characters (14 for
the salt plus the key). If different ciphers are chosen for RTP and
RTCP the maximum needed key size is expected.
* ext/srtp/gstsrtpdec.c: minor documentation updates.
https://bugzilla.gnome.org/show_bug.cgi?id=720434
Alternates between 33 and 32 byte frames, but must start
with a 33 byte frame. This has been broken for ages since
the element was ported to the audio decoder base class.
https://bugzilla.gnome.org/show_bug.cgi?id=709416
This currently converts from ARGB64_F16 (16 bit float per component)
to ARGB64 by clipping. We should add support for the F16 format and
implement a conversion filter element that can apply gamma curves,
change exposure, etc.
It only gets the sink flag set when it adds the multifilesink, that
happens in null->ready and it might be too late. Set the flag
explicitly on the constructor.
https://bugzilla.gnome.org/show_bug.cgi?id=711086
This patch fixes three memory leaks in hlsdemux, one that occurs
during normal operation and two that occur during error conditions.
The gst_hls_demux_get_next_fragment function calls
gst_fragment_get_buffer which increments the reference count
on the buffer but gst_hls_demux_get_next_fragment never calls unref on
the buffer. This means that the reference count for each downloaded
fragment never gets to zero and so its memory is never released.
This patch adds a call to gst_buffer_unref after the flags have been
updated on the buffer.
There is a leak-on-error in gst_hls_demux_decrypt_fragment if it fails
to download the key file. If the key fails to download, null is
returned without doing an unref on the encrypted fragment. The
semantics of gst_hls_demux_decrypt_fragment is that it takes ownership
of the encrypted fragment and releases it before returning.
There is a leak-on-error in gst_hls_src_buf_to_utf8_playlist in the
unlikely event that the gst_buffer_map fails. In the "happy path"
operation of gst_hls_src_buf_to_utf8_playlist the buffer gets an unref
before the function returns, therefore the error condition must do the
same.
https://bugzilla.gnome.org/show_bug.cgi?id=710881
Fixed up the error-handling code when downloading fragments.
Modifed the error-handling code to use positive logic when
testing for cancellation of the download loop.
https://bugzilla.gnome.org/show_bug.cgi?id=701404
There is an issue for live streams where download_loop will keep
downloading segments until it gets a 404 error for a segment
that has not yet been published. This is a problem because this
request for a segment that doesn't exist will propagate all the
way back to the origin server(s). This means that dashdemux causes
extra load on the origin server(s) for segments that aren't yet
available.
This patch uses availabilityStartTime, period
and the host's idea of UTC to decide if a fragment is available to
be requested from an HTTP server and filter out requests for fragments
that are not yet available.
https://bugzilla.gnome.org/show_bug.cgi?id=701404
On some live HLS streams, gst_hls_demux_switch_playlist causes
assertion failures because it tried to dereference a NULL fragment.
This is because g_queue_peek_tail sometimes was returning NULL and
this case was not being checked.
This patch does two things:
* move the g_queue_peek_tail inside the semaphore protection
* check if q_queue_peek_tail returns NULL
https://bugzilla.gnome.org/show_bug.cgi?id=708849
gstdashdemux.c:1753: warning: format '%llu' expects type 'long long unsigned int', but argument 8 has type 'long unsigned int'
gstdashdemux.c:2224: warning: format '%llu' expects type 'long long unsigned int', but argument 9 has type 'guint64'
gstdashdemux.c:2224: warning: format '%llu' expects type 'long long unsigned int', but argument 10 has type 'guint64'
gstmpdparser.h:530: warning: type qualifiers ignored on function return type
gstmpdparser.c:4177: warning: type qualifiers ignored on function return type
including the following supports and fixes:
* Create DirectFB surfaces from GstBufferPool
* Add NV12 pixel format support
* Don't use the cursor in the exclusive mode
- EnableCusor() can be only used when the administrative mode is set
in DirectFB 1.6.0 and later.
* Support multiple plane rendering for planar color formats
- This accommodates the chroma plane offsets of the framebuffer
in planar formats.
* Invoke SetConfiguration regardless of video mode setting in setcaps()
- SetConfiguration() method should be invoked regardless of
the result of gst_dfbvideosink_get_best_vmode(), since the two are
unrelated.
* Disable DirectFB signal handler
- "--dfb:no-sighandler" option is passed to DirectFBInit().
This prevents DirectFB from trying to kill the process and allows
GStreamer's termination sequence to proceed normally.
https://bugzilla.gnome.org/show_bug.cgi?id=703520
For SegmentTemplate elements containing a startNumber attribute, the
`number' member of GstMediaSegments should be offset by the value of
startNumber; however, this is not currently the case. As a result, the
first URI(s) requested by the download loop will be wrong.
This commit ensures that segment numbers will be offset by startNumber
when one is present in a SegmentTemplate element.
https://bugzilla.gnome.org/show_bug.cgi?id=705661
When using a SegmentTemplate element, the timestamps of the buffers
output by dashdemux are incorrect, causing problems downstream.
The reason is that GstMediaSegment start times are calculated (in
gst_mpdparser_get_chunk_by_index) by multiplying segment index by
segment duration and then scaling the result according the `timebase'
attribute from the MPD. However, the segment duration is already a
GstClockTime (i.e., it has already been scaled according to the timebase
from the MPD and converted to a nanosecond value), so multiplying it by
the segment index will give the correct timestamp without the need for
any further scaling.
https://bugzilla.gnome.org/show_bug.cgi?id=705679
This prevents locking on startup when a stream only has a single buffer
for one of the streams and mssdemux decides to push an EOS event right
after it.
This prevents deadlocks on startup on files that have only a very
large buffer for a stream and the queue is filled and will lock on
the eos event that is pushed after the buffer. As no buffers have yet
been pushed to other streams, the pipeline locks on preroll
Every encrypted fragment will be a multiple of 128 bits, the last byte
contains the number of bytes that were added as padding in the end
and should be removed.
https://bugzilla.gnome.org/show_bug.cgi?id=701673
When using an HLS encrypted stream, an assertion failure is thrown:
(gst-launch-1.0:31028): GLib-GObject-WARNING **: cannot register
existing type `GstFragment'
(gst-launch-1.0:31028): GLib-CRITICAL **: g_once_init_leave: assertion
`result != 0' failed
Eventually tracked this down to the call gst_fragment_new()
in function gst_hls_demux_decrypt_fragment.
The GstFragment class is defined in ext/hls/gstfragment.c and in
gst-libs/gst/uridownloader/gstfragment.c. Having two class definitions
with the same name causes the assert failure when trying to allocate
GstFragment. Deleting the version from hls and editing the
Makefile.am solves this assert failure.
https://bugzilla.gnome.org/show_bug.cgi?id=704555
During a live stream it is possible for dashdemux to lag behind on a
slow connection or to rush ahead of the connection os too fast.
For the first case it is necessary to jump some segments ahead to be able to
continue playback as old segments are usually deleted from the server.
For the later, dashdemux should wait a little before attempting another
download do give time to the server to produce a new segment
When using a template based segment list, do not try to
contruct a finite segment list for the limits of the available periods.
We might not know when the period ends (for live streams) and we can
always create the segment on demand when requested by dashdemux,
avoiding use of some memory and cpu when re-creating this list.
Replaces the 2 likely larger lists with more appropriate structures
to improve performance.
Replaces S nodes GList for a GQueue, this reduces latency to startup
because of traversing the list just append an element.
Replaces the processed media segments GList for a GPtrArray as it is
constantly acessed by index during playback.
Duration from segment being unknown is a issue from the MPD and not
a programming issue, so the assert isn't useful here. Instead check
and return an error code so the caller can fallback to alternatives
When dashdemux selects its first fragment, it always selects the
first fragment listed in the manifest. For on-demand content,
this is the correct behaviour. However for live content, this
behaviour is undesirable because the first fragment listed in the
manifest might be some considerable time behind "now".
The commit uses the host's idea of UTC and tries to find the
oldest fragment that contains samples for this time of day.
https://bugzilla.gnome.org/show_bug.cgi?id=701509
According to the MPEG-DASH spec, certain elements (i.e.
SegmentBase, SegmentTemplate, and SegmentList) should inherit
attributes from the same elements in the containing AdaptationSet
or Period.
Updated the SegmentBase, SegmentTemplate, and SegmentList parsers
to properly inherit attributes from the corresponding elements in
AdaptationSet and/or Period.
https://bugzilla.gnome.org/show_bug.cgi?id=702677
Convert all xml attribute/content parsing functions to return a
boolean value indicating whether or not the attribute/content was
present. We need this finer-grained control in order to properly
implement the inheritance policies described in the spec
Also fixed several memory leak conditions when handling errors in
the xml attribute/content parsing functions.
https://bugzilla.gnome.org/show_bug.cgi?id=702677
Ensure that g_free/xmlFree is used correctly based on how the
memory was allocated.
When deallocating GLists, there were many places that were using
g_list_foreach and g_list_free. Converted these occurrences to
call g_list_free_full.
Add NULL checks to all xmlFree calls since the documentation does
not guarantee that passing NULL is safe
In places where we are strdup'ing memory allocated by libxml2,
changed those calls to use xmlMemStrdup().
There were several places where we were missing g_slice_free when
deallocating a top-level node structure.
https://bugzilla.gnome.org/show_bug.cgi?id=702837
Wayland interface could offer two buffers pixels formats: WL_SHM_FORMAT_XRGB8888 and WL_SHM_FORMAT_ARGB8888.
Update waylandsink to support them and check if the format is really available.
https://bugzilla.gnome.org/show_bug.cgi?id=702112
Fixes:
In file included from gstsegmentation.h:51:0,
from gstopencv.c:42:
/usr/include/opencv2/video/background_segm.hpp:47:16: fatal error: list:
No such file or directory
#include <list>
^
compilation terminated.
https://bugzilla.gnome.org/show_bug.cgi?id=702297
It was not properly divided by GST_SECONDS. Also fix issue with
max-buffering-time being multiplied by GST_SECONDS every time the
property is retrieved.
https://bugzilla.gnome.org/show_bug.cgi?id=700487
Split the introspection and registration part. This way we only need to open all
plugins when updating the registry. When reading the registry we can register
the elements entierly from the cache.
Add colour image enhancement element based on Retinex algorithm. Two types
exist, namely basic and multiscale; both are described in this article:
Rahman, Zia-ur, Daniel J. Jobson, and Glenn A. Woodell. "Multi-scale retinex
for color image enhancement." Image Processing, 1996. Proceedings.,
International Conference on. Vol. 3. IEEE, 1996
Visually speaking the result looks a bit funny, but is pretty invariable to
lightning changes, which is good for some applications, like image
segmentation.
https://bugzilla.gnome.org/show_bug.cgi?id=700977
WmaPro is actually wmaversion 3, and can also be found by the
WMAP fourcc.
Some manifests also contain the block_align field as "PacketSize"
in the audio track description, the libav decoders require it
to be present in caps.
Fixes#699921
Detect when the eagl surface changed its dimension (when the user rotates
the device for example) and adapt the egl internals to draw to that,
preventing that ios resizes the image again when drawing.
This is particularly harmful when eagl would scale down a image
to draw and the ios screen would scale it back up because the
surface is now bigger than when the element was configured.
wma v2 expects block_align, channels and rate fields set to its caps.
This isn't present direclty on the manifests, so mssdemux should parse
it from the waveformatex structure
https://bugzilla.gnome.org/show_bug.cgi?id=699924
bitrate info is always present on the QualityLevel xml node as part
of the adaptive selection processing, put it into caps as some
decoders require it (avdec_wmav2 for example)
https://bugzilla.gnome.org/show_bug.cgi?id=699924
It's not developed any more and replaced by the
libschroedinger-based elements in gst-plugins-good.
(The libschroedinger 1.0.9 release notes state "This
is an exciting release: most of the encoding tools in
dirac-research have been ported over to Schrödinger, so
now schro has the same or better compression efficiency
as dirac-research.")
TRM IDs are MusicBrainz' old audio fingerprinting system from
Relatable, they were phased out in favour of MusicIPs PUIDs.
https://wiki.musicbrainz.org/History:TRM
In some scenarios, for example in QtWebKit, might be difficult to obtain full
control on the egl display and it might be only accessible indirectly via
eglGetCurrentDisplay().
https://bugzilla.gnome.org/show_bug.cgi?id=700058
We only want to adjust the timestamps so that they start from 0 for live
streams. Non-live streams already start from 0 and after a seek we actually want
to timestamp to be the position we seek to.
Non-live streams should timestamp buffers with a running-time starting from
0. Since we already push a 0 -> -1 segment, bring the timestamps to 0
by subtracting the initial timestamp.
The xmlCleanupParser function seems to cleanup all statically
allocated libxml variables, making it unusable. We can't guarantee
that dashdemux won't need it anymore, so better not call it.
Manifest updates should be done periodically for live streams,
this patch makes the demuxer create a new manifest client for
the new version and transfers the stream position to the new
one, discarding the old one afterwards.
A small struct that keeps a short history of fragment download bitrates
to have an average measure of N last fragments instead of using only
the last downloaded bitrate
Do not use a global bitrate as the sizes of the fragments matter
when calculating the download rate as the connection setup time is
also being taken into the download duration, a smaller fragment
will have a lower bitrate than a larger one.
This avoids switching the bitrates for streams frequently because
of bitrate mismatches
Instead of downloading 1 fragment per stream per download loop,
select the stream with the earlier timestamp and get a fragment
only for that one.
The old algorithm would lead to problems when the fragment durations
were too different for streams.
dashdemux shouldn't emit the buffering message as that can pause
the pipeline. It has no proper knowledge of the downstream buffering
status so it can pause the pipeline when it isn't necessary. It should
have an internal buffer for downloading the streams ahead of playback,
but that shouldn't make it able to stop the pipeline for buffering.
A particular case in which this is bad is when a pad switch happens
(changing bitrates for example), the new pads dashdemux creates
will get linked to demuxers and new queues will be created,
these queues are initially empty and dashdemux will quickly
drain its buffers by pushing them to those queues. So it
would have no more buffers internally and would emit a
buffering message with a low ratio, causing the pipeline
to pause when it wouldn't be necessary.
Put EOS on the streams queues after the last fragment from the
last period for each stream. This way we keep it serialized
with the buffers and it will work when streams have different
ending times
The smallest queue should be used to prevent blocking the download
thread when a stream has too much data buffered, leaving the other
streams starving from fragments
Each stream has its own durations and timestamps, the fragment number
is different for each stream when seeking, so the seek has to be done
for all streams, rather than on a single stream and propagated to
others
GstDataQueue has proper locking and provides functions to limit the
size of the queue. Also has blocking calls that are useful to
our multithread scenario in Dash.
Store the buffers separately for each stream, this is clearer than
having a queue with a list of buffers. It also allows easier selection
of buffers to push in later refactors
Fragments should be pushed ASAP as downstream should be responsible for
doing the syncrhonization and proper buffering.
This has the great side effect of fixing most of the seeking A/V sync issues.
- the MPD file is updated in the download loop (only if we have a "dynamic" MPD and minimumUpdatePeriod is valid);
- properly LOCK/UNLOCK the GstMpdClient;
This fixes conflicts with the HLS plugin, which is also named
fragmented.
When building its registry, gstreamer was picking one or the other
between hls and dashdemux.
This fixes build that has been broken by commit
fb9aeac6552021b176a4c4bd07265e02a0b70e0f.
gst_mpd_client_get_target_duration has been removed, and
gst_mpd_client_get_next_fragment_duration should be used instead.
This was necessary to support variable-duration Fragments.
in the new API:
- gst_mpd_client_get_current_position returns the timestamp of the NEXT fragment to download;
- gst_mpd_client_get_next_fragment_duration returns the duration of the next fragment to download;
- gst_mpd_client_get_media_presentation_duration returns the mediaPresentationDuration from the MPD file;
also there is a new internal parser function:
- gst_mpd_client_get_segment_duration extracts the constant segment duration from the MPD file
(only used when there is no SegmentTimeline syntax element in the current representation)
In gst_mpd_client_get_next_fragment, we set the timestamp/duration of the fragment just downloaded
copying the values from the corresponding GstMediaSegment.
TODO: rework SEEKING to support seeking across different Periods.
- Periods are played in sequence, from PeriodStart to PeriodEnd
- seamless switching from one Period to the next one works fine;
- the 'new-segment' generation is broken, so if we need to switch pads for a new Period there is a crash;
- build a list of the available Periods with their start and duration time
- add the list of GstStreamPeriod in the GstMpdClient data struct
- remove cur_period from GstMpdClient and introduce an API to get the current GstStreamPeriod
- several API clean-ups
build the list of segments to be played using the SegmentTimeline syntax, if present
bugfixes:
- for dynamic MPD files, when mediaPresentationDuration is not present use minimumUpdatePeriod instead
- do not add a spurious '$' when building an URL from a template like "$Bandwidth$/init.mp4v"
- introduce gst_mpd_client_add_media_segment() to avoid code duplication
other fixes:
- fixed a buffering bug: now we stop buffering when we reach the end of manifest
- now gst_mpd_client_get_target_duration() always returns a valid duration
(in case of single-segment streams, we return either Period duration or mediaPresentation duration)
TODO: support SegmentTimeline
SegmentList nodes are allowed into Period, AdaptationSet or Representation nodes
and there is at most 1 element, so no need to keep a list;
Period nodes cannot have any Represention elements, as AdaptationSet nodes are mandatory;
this breaks compatibility with some legacy DASH test sequences.
gstmpdparser.c: In function ‘gst_mpdparser_get_list_and_nb_of_audio_language’:
gstmpdparser.c:2891: warning: ‘return’ with no value, in function returning non-void
g_ascii_strtoull() returns a long long integer, but we need to
pass a normal int to gst_structure_set() for fields of G_TYPE_INT,
so cast appropriately.
The buffer parameter wasn't being used, it was only to signal if
a buffer was downloaded and advance to the next fragment in the
manifest.
Replace the buffer with a boolean that has the same effect and is
safer
connection setup times seem to matter when measuring the download
rate of different streams. Streams with longer fragments have a
*relatively* lower connection setup time and achieve higher bitrates.
Using the average seems unfair here, so use each stream's measured bitrate
to select its best quality option.
We need to cancel the downloader for each stream before joining the main download task, otherwise
the download task will block until all the stream tasks finish.
When the codec is AAC-LC, some server implementation (e.g. Microsoft IIS) doesn't add the CodecPrivateData attribute. The element needs to re-create the codec data from the Quality Level attributes (channels and sampling rate).
There is no way to know if a live stream is really finished, so try to reload the manifest and check if there are more fragments to download. Else just let know it's the EOS.
Live streams force the demuxer to keep reloading the Manifest from
time to time, as the new fragments are being added as they are recorded.
The demuxer should also try to keep up and detect when it had to skip
fragments, marking the discont flag when that happens.
Curiously, the spec doesn't seem to mention when/how a live stream is supposed
to end, so keep trying downloads until the demuxer errors out.
Use pad tasks to download data and an extra task that gets the earlier
buffer (with the smallest timestamp) and pushes on the corresponding
pad.
This prevents that the audio stream rushes ahead on buffers as its
fragments should be smaller
When connection-speed changes, signal that we might need a bitrate
switch. During the switch, a new pad group is added and the old one
is drained and removed.
New pads also need to push newsegments before starting to stream
This speed limits the maximum bitrate of streams. Currently it
is only read when starting the pipeline, but it should be used
for switching bitrates during playback to adapt to network
changes.
mssdemux should set the streams it has exposed as active so that
the manifest won't use the non-active streams to compute total bitrates
or providing fragments
When the stream can't have its caps detected, better not to expose it.
If no streams are known, signal an error about no playable streams to
the application
Adds basic handling for seek in time events. Needs to cooperate
with the downstream qtdemux so that it forwards the seeks and
the corresponding newsegments
This is important for downstream to properly timestamp the samples
The default value is 10000000, but this can be set in the stream
or at the top-level manifest entry
Keep a ref on pad to prevent it being unreffed while the mssdemux
streams are still using it. Also reset the element when going to
ready instead of when going to null.
Use shorter names for the MSS manifest helper structure and functions.
Also continues the implementation of the stream fetching and pushing loop.
Now it uses the base url correctly and already fetches and pushes the fragments
downstream
It seems EAGL expects the application to simply ignore unused
EAGL contexts as the resources for it would be released when a new
context is set as the current one. Also move the egl extensions
querying to after a context is set to prevent crashes.
This makes the EAGL version of eglglessink reusable.
gstegladaptation_egl.c: In function 'gst_egl_adaptation_create_native_window':
gstegladaptation_egl.c:868:3: error: format '%p' expects argument of type 'void *', but argument 8 has type 'EGLNativeWindowType' [-Werror=format=]
GST_DEBUG_OBJECT (ctx->element, "Using window handle %p", window);
^
Put EGL specific code to a separate file and create the same functions
for EAGL, the Apple's specific EGL implementaton.
At this point, the EAGL version wasn't compiled or tested as there isn't
any simple documented way to build 1.0 for iOS. This code for the EAGL
version is still the 0.10 version, some updates should be made when 1.0
is buildable for iOS.
gsteglglessink.c: In function 'gst_eglglessink_fill_texture':
gsteglglessink.c:1815:3: error: format '%d' expects argument of type 'int', but argument 11 has type 'gsize' [-Werror=format]
gsteglglessink.c: In function 'gst_eglglessink_configure_caps':
gsteglglessink.c:2850:3: error: format '%p' expects argument of type 'void *', but argument 8 has type 'EGLNativeWindowType' [-Werror=format]
