When start qmlglsink app, it will set NULL buffer to GstQSGTexture
in which case that qt_context_ will be a random value and cause
gst_gl_context_activate() fail.
https://bugzilla.gnome.org/show_bug.cgi?id=770925
Previously it was created in the init function and destroyed in ::stop, which
lead to segfaults when reusing the element.
Now the upload object is created in ::transform_caps if it is NULL, which is the
earliest we need it. The other vfuncs already bail out if the upload object is
NULL, which means that negotiation wasn't done.
Now when used with video/x-raw as input, the GLMemoryUpload method checks for
->tex_target in input GLMemory(es) and sets the output texture-target
accordingly.
Fixes video corruption with a pipeline like avfvideosrc ! video/x-raw !
glimagesink where on macos avfvideosrc pushes RECTANGLE textures but glupload
was configuring texture-target=2D as output.
The videoaggregator negotiation sequence changed some time
back and broke glstereomix. Instead of doing nego incorrectly
in the find_best_format() vfunc, do it directly in the
update_caps() method.
And scale the bitrate with the absolute rate (if it's bigger than 1.0) to get
to the real bitrate due to faster playback.
This allowed in my tests to play a stream with 10x speed without buffering as
the lowest bitrate is chosen, instead of staying/selecting the highest bitrate
and then buffering all the time.
It was previously disabled for not very well specified reasons, which seem to
be not valid anymore nowadays.
It implements now this interface with its video-direction
property. Values are changed to GstVideoOrientationMethod but they have
the same value than the originals.
https://bugzilla.gnome.org/show_bug.cgi?id=768687
https://github.com/mesonbuild/meson
With contributions from:
Tim-Philipp Müller <tim@centricular.com>
Matej Knopp <matej.knopp@gmail.com>
Jussi Pakkanen <jpakkane@gmail.com> (original port)
Highlights of the features provided are:
* Faster builds on Linux (~40-50% faster)
* The ability to build with MSVC on Windows
* Generate Visual Studio project files
* Generate XCode project files
* Much faster builds on Windows (on-par with Linux)
* Seriously fast configure and building on embedded
... and many more. For more details see:
http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.htmlhttp://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html
Building with Meson should work on both Linux and Windows, but may
need a few more tweaks on other operating systems.
_stdint.h is generated by Autotools and we don't really need it. All
supported platforms now ship with stdint.h. The only stickler was MSVC,
and since Visual Studio 2015 it also ships stdint.h now.
It now returns the correct values for both orthographic and perspective
projections and takes into account the aspect ratio of the video, handles
the Y-flipping in GL and by us and uses some more helpers from graphene.
Fixes spurious segfault in unit test, where the task was started again during
shutdown when all pads were removed... and was then still running while the
element was finalized.
We don't have to do yet another additional request but can just download the
data directly.
Also unify the key-unit only mode buffer pushing and extract it into its own
function now that it became more complicated.
https://bugzilla.gnome.org/show_bug.cgi?id=741104
We need to mark every first buffer of a key unit as discont, and also every
first buffer of a moov and moof. This ensures that qtdemux takes note of our
buffer offsets for each of those buffers instead of keeping track of them
itself from the first buffer. We need offsets to be consistent between moof
and mdat
https://bugzilla.gnome.org/show_bug.cgi?id=741104
Fixes the following error when building in osx.
error: implicit conversion from enumeration type
'GstJPEG2000Colorspace' to different enumeration type
'GstJPEG2000Sampling'
This reverts commit 947656cfd2.
This makes all dash seeking tests fail. Needs more testing to fully understand
what's going wrong. Revert ok'd by Sebastian
We don't need to call the latter at all as we're definitely in this period and
the segment is selected via the SIDX.
This is especially important when doing SNAP seeks, as otherwise we would
always start from the beginning of the period (usually 0) again.
After the check in line 1,111, media->uri can't be NULL. So the two checks
for GST_HLS_MEDIA_TYPE_CLOSED_CAPTIONS are the same, removing the redundant
one which goes to cc_unsupported.
CID 1364752
Create an output stream for each media when alternate renditions
are present. Update the manifests for all those streams, and
make sure that typefinding is still done for files smaller than 2KB
such as small WebVTT files.
When fetching a byte-region from a server resource,
adjust the downstream buffer offsets so that downstream
doesn't know. This is because id3demux insists on the
first offset being 0. Later we might strip ID3 headers
entirely and this will be unneeded.
Modify playlist updating to track information across updates
better, although still hackish.
When connection_speed == 0, choose the default variant
not the first one in the (now sorted) variant list, as that
will have the lowest bitrate.
Make M3U8 and GstM3U8MediaFile refcounted. The contents
of it and GstM3U8MediaFile are pretty much immutable
already, but if we make it refcounted we can just
return a ref to the media file from _get_next_fragment()
instead of copying over all fields one-by-one, and then
copying them all into the adaptive stream structure fields again.
Move state from client into m3u8 structure. This will
be useful later when we'll have multiple media playlists
being streamed at the same time, as will be the case with
alternative renditions.
This has the downside that we need to copy over some
state when we switch between variant streams.
The GstM3U8Client structure is gone, and main/current
lists are not directly in hlsdemux. hlsdemux had as
many CLIENT_LOCK/UNLOCK as the m3u8 code anyway...
The gst_dash_demux_get_live_seek_range () function returns a stop value
that is beyond the available range. The functions
gst_mpd_client_check_time_position() and
gst_mpd_client_get_next_segment_availability_end_time() in
gstmpdparser.c include the segment duration when checking if a segment
is available. The gst_dash_demux_get_live_seek_range() function
in gstdashdemux.c ignores the segment duration.
According to the DASH specification, if maxSegmentDuration is not present,
then the maximum Segment duration is the maximum duration of any Segment
documented in the MPD.
https://bugzilla.gnome.org/show_bug.cgi?id=753751
There's no need for the jump to an extra thread in most cases, especially
when relying solely on a shader to render. We can use the provided
render_to_target() functions to simplify filter writing.
Facilities are given to create fbo's and attach GL memory (renderbuffers
or textures). It also keeps track of the renderable size for use with
effective use with glViewport().
Don't clear decryption state immediately after
initialising it in the start_fragment. Don't clear
the state of all streams when we want to only clear
the current stream.
https://bugzilla.gnome.org//show_bug.cgi?id=768757
Add demuxer instance-wide decryption key cache. The current and
last key url are per-stream, so make a shared cache. Move the
decryption handling into the stream object, and use the shared
cache for the keys.
Prepare hlsdemux for more than one single stream. Currently hlsdemux
assumes there'll only ever be one stream and most of the stream-specific
state is actually in the hlsdemux structure. Add a stream subclass
instead and move some stream-specific members there instead.
In this mode, we let WebRTC Audio Processing figure-out the delay. This
is useful when the latency reported by the stack cannot be trusted. Note
that in this mode, the leaking of echo during packet lost is much worst.
It is recommanded to use PLC (e.g. spanplc, or opus built-in plc).
In this mode, we don't do any synchronization. Instead, we simply process all
the available reverse stream data as it comes.
Compiler would complain about include directory that didn't
exist because QPA_INCLUDE_PATH gets subst-ed regardless
(and if it didn't we'd have just an empty -I argument).
https://bugzilla.gnome.org/show_bug.cgi?id=767553
This simplifies the code but also removes a bug with tracking of the remaining
size for the initial subfragment: we were not considering the size between the
index and the start of the first moof here.
https://bugzilla.gnome.org/show_bug.cgi?id=764684
When switching fragments we don't want to keep any data around from the last
one, and also forget about all data when doing flushing seeks or selecting new
bitrates.
https://bugzilla.gnome.org/show_bug.cgi?id=764684
The previous code would run out of sync if there was packet lost
or clock skews. When that happened, the echo cancellation feature would
completely stop working. As this is crucial for audio calls, this patch
re-implement synchronization completely.
Instead of letting it drift until next discont, we now synchronize
against the record data at every iteration. This way we simply never
let the stream drift for longer then 10ms period. We also shorter the
delay by using the latency up the probe (basically excluding the sink
latency. This is a decent delay to avoid starving in the probe queue.
https://bugzilla.gnome.org/show_bug.cgi?id=768009
When echo cancel is enabled, we now fail the pipeline if there is
not echo probe. For this reason there is no need to check if probe
pointer is set anymore.
The byte-stream to avc conversion did not consider NAL sizes bigger than 2^16,
multiple layers, multiple NALs per layer, and various other things. This
caused corrupted streams in higher bitrates and other circumstances.
Let's just forward byte-stream as generated by the encoder and let h264parse
handle conversion to avc if needed. That way we only have to keep around one
version of the conversion and don't have to fix it in multiple places.
Rather than assuming something. e.g. zerocopy on iOS with GLES3 requires
the use of Luminance/Luminance Alpha formats and does not work with
Red/RG textures.
The saved timestamp is used to compute the delay of the probe data.
As it's used at the following incoming buffer, it needs to be offset
with the duration of the buffer to represent the end position. Also,
properly initialize the saved timestamp and protect against TIME_NONE.
Until now, we were synchronizing both DSP and Probe adapter by
waiting and clipping the probe adapter data. This increases the CPU
usage, can cause copies if the audio is not 10ms aligned and the worst
is that it prevents the processing from compensating for inaccurate
latency. This is also a step forward toward supporting playback
filters.
The current state of c++ ABI's on Window's and Gst's/Qt's conflicting
mingw builds means that we cannot use mingw for building the qt plugin.
Instead, a qmake .pro file is provided that is expected to be used with the
msvc binaries provided by Qt like so:
(with the PATH environment variable containing the path to the qt biniaries
and PKG_CONFIG_PATH containing the path to GStreamer modules)
cd /path/to/sources/gst-plugins-bad/ext/qt
qmake -tp vc
Then open the resulting VS project and build the library. Then
cp debug/libgstqtsink.dll /path/to/prefix/lib/gstreamer-1.0/libgstqtsink.cll
https://bugzilla.gnome.org/show_bug.cgi?id=761260
This DSP library can be used to enhance voice signal for real time
communication call. In implements multiple filters like noise reduction,
high pass filter, echo cancellation, automatic gain control, etc.
The webrtcdsp element can be used along, or with the help of the
webrtcechoprobe if echo cancellation is enabled. The echo probe should
be placed as close as possible to the audio sink, while the DSP is
generally place close to the audio capture. For local testing, one can
use an echo loop pipeline like the following:
autoaudiosrc ! webrtcdsp ! webrtcechoprobe ! autoaudiosink
This pipeline should produce a single echo rather then repeated echo.
Those elements works if they are placed in the same top level pipeline.
https://bugzilla.gnome.org/show_bug.cgi?id=767800
These two shader will calculate the vector length and use it as denominator.
But length could be zero which will cause undefine behaviour. Add protection for
this condition
https://bugzilla.gnome.org/show_bug.cgi?id=767635
We remove get-rollover-counter signal in favor of the "stats"
property. The "stats" property is a GstStructure with caps
application/x-srtp-encoder-stats that contains an array of
structures with caps application/x-srtp-stream.
Each stream structure contains "ssrc" and "roc" fields.
https://bugzilla.gnome.org/show_bug.cgi?id=733265
Previously GstCurlSmtpSink could cause the pipeline thread to end up
waiting for a stopped thread to perform work.
The scenario was that the sink could be rendering a buffer and waiting
for the curl transfer thread to have sent the data. As soon as the
transfer thread has copied all data to curl's data buffer in
gst_curl_base_sink_transfer_read_cb() then the render call would stop
waiting and return GST_FLOW_OK. While this takes place the transfer
thread may suffer from an error e.g. due gst_poll_wait() timing out.
This causes the transfer thread to record the error, claim (it is not
really true since there was an error) that the data has been sent and
that a response has been received by trying to signal the pipeline
thread (but this has already stopped waiting). Finally the transfer
thread stops itself. A short while later the pipeline thread may attempt
to push an EOS event into GstCurlSmtpSink. Since there is no check in
gst_curl_smtp_sink_event() to check if the sink has suffered from any
error it may attempt to add a final boundary and ask the, now deceased,
transfer thread to transfer the new data. Next the sink element would
have waited for the transfer to complete (using a different mechanism
than normal transfers through GstCurlBaseSink). In this case there was
an error check to avoid waiting if an error had already been seen.
Finally GstCurlSmtpSink would chain up to GstCurlBaseSink which would
then block waiting for a response (normally this would be prevented by
the transfer thread suffering the error claiming that it had been
received, but GstCurlSmtpSink clobbered this flag after the fact).
Now GstCurlSmtpSink avoids this by locking over the entire event handing
(preventing simultaneous changes to flags by the two threads) and also
by avoiding to initiate transfer of final boundary if an error has
already been seen.
Also add GST_FIXME() for remaining similar issue where the pipeline
thread may block indefinitely waiting for transfer thread to transfer
data but the transfer thread errors out and fails to notify the pipeline
thread that the transfer failed.
https://bugzilla.gnome.org/show_bug.cgi?id=767501
Use namespace only after it was actually defined by a header.
gstfacedetect.cpp:79:17: error: using directive refers to implicitly-defined namespace 'std' [-Werror]
using namespace std;
^
Replace the type and function prefix to follow the conventions:
- Use `GST_TYPE_DC1394_SRC` instead of `GST_TYPE_DC1394`.
- Use `GstDC1394Src` and `GstDC1394SrcClass` instead of
`GstDc1394` and `GstDc1394Class`.
- Use `gst_dc1394_src` instead of `gst_dc1394`.
https://bugzilla.gnome.org/show_bug.cgi?id=763026
The dc1394src is a PushSrc element for IIDC cameras based on libdc1394.
The implementation from the 0.x series is deffective:
caps negotiation does not work, and some video formats
provided by the camera are not supported.
Refactor the code to port it to 1.X and enhance the support
for the full set of video options of IIDC cameras:
- The IIDC specification includes a set of camera video modes
(video format, frame size, and frame rates).
They do not map perfectly to Gstreamer formats, but those that
do not match are very rare (if used at all by any camera).
In addition, although the specification includes a raw format,
some cameras use mono video formats to capture in Bayer format.
Map corresponding video modes to Gstreamer formats in capabilities,
allowing both gray raw and Bayer video formats for mono video modes.
- The specification includes scalable video modes (Format7),
where the frame size and rate can be set to arbitrary values
(within the limits of the camera and the bus transport).
Allow the use of such mode, using the frame size and rate
from the negotiatied caps, and set the camera frame rate
adjusting the packet size as in:
<http://damien.douxchamps.net/ieee1394/libdc1394/faq/#How_do_I_set_the_frame_rate>
The scalable modes also allow for a custom ROI offset.
Support for it can be easily added later using properties.
- Camera operation using libdc1394 is as follows:
1. Enumerate cameras on the system and open the camera
identified the enumeration index or by a GUID (64bit hex code).
2. Query the video formats supported by the camera.
3. Configure the camera for the desired video format.
4. Setup the capture resources for the configured video format
and start the camera transmission.
5. Capture frames from the camera and release them when not used.
6. Stop the camera transmission and clear the capture resources.
7. Close the camera freeing its resources.
Do steps 2 and 3 when getting and setting the caps respectively.
Ideally 4 and 6 would be done when going from PAUSED to PLAYING
and viceversa, but since caps might not be set yet, the video mode
is not properly configured leaving the camera in a broken state.
Hence, setup capture and start transmission in the set caps method,
and consequently clear the capture and stop the transmission
when going from PAUSED to READY (instead of PLAYING to PAUSED).
Symmetrycally, open the camera when going from READY to PAUSED,
allowing to probe the camera caps in the negotiation stage.
Implement that using the `start` and `stop` methods of `GstBaseSrc`,
instead of the `change-state` method of `GstElement`.
Stop the camera before setting new caps and restarting it again
to handle caps reconfiguration while in PLAYING (it has no effect
if the camera is not started).
- Create buffers copying the bytes of the captured frames.
Alternatively, the buffers could just wrap the bytes of the frames,
releasing the frame in the buffer's destroy notify function,
if all buffers were destroyed before going from PLAYING to PAUSED.
- No timestamp nor offset is set when creating buffers.
Timestamping is delegated to the parent class BaseSrc,
setting `gst_base_src_set_live` TRUE, `gst_base_src_set_format`
with GST_FORMAT_TIME and `gst_base_src_set_do_timestamp`.
Captured frames have a timestamp field with the system time
at the completion of the transmission of the frame,
but it is not sure that this comes from a monotonic clock,
and it seems to be left NULL in Windows.
- Use GUID and unit properties to select the camera to operate on.
The camera number used in version 0.X does not uniquely identify
the device (it depends on the set of cameras currently detected).
Since the GUID is 64bit identifier (same as MAC address),
handle it with a string property with its hexadecimal representation.
For practicality, operate on the first camera available if the GUID
is null (default) and match any camera unit number if unit is -1.
Alternatively, the GUID could be handed with an unsigned 64 bit
integer type property, using `0xffffffffffffffff` as default value
to select the first camera available (it is not a valid GUID value).
- Keep name `GstDc1394` and prefix `gst_dc1394` as in version 0.X,
although `GstDC1394Src` and `gst_dc1394_src` are more descriptive.
- Adjust build files to reenable the compilation of the plugin.
Remove dc1394 from the list of unported plugins in configure.ac.
Add the missing flags and libraries to Makefile.
Use `$()` for variable substitution, as many plugins do,
although other plugins use `@@` instead.
https://bugzilla.gnome.org/show_bug.cgi?id=763026
The heuristic to choose between packetise or not was changed to use the
segment format. The problem is that this change is reading the segment
during the caps event handling. The segment event will only be sent
after. That prevented the decoder to go in packetize mode, and avoid
useless parsing.
https://bugzilla.gnome.org/show_bug.cgi?id=736252
Use new gst_h264_video_calculate_framerate() API instead of fps_n/fps_d
fields in SPS struct which are to be removed.
Apparently H264 content in MSS is always non-interlaced/progressive,
so we can just pass 0 for field_pic_flag and don't need to parse any
slice headers first if there's no external signalling. But even if
that's not the case the new code is not worse than the existing code.
https://msdn.microsoft.com/en-us/library/cc189080%28VS.95%29.aspxhttps://bugzilla.gnome.org/show_bug.cgi?id=723352
_get_gl_context() can be called concurrently from either propose_allocation() or
decide_allocation(). If it so happens that this happens at the same time,
the check for whether we already had a GL context was outside the lock. Inside
the lock and loop, the first thing that happens is that we unref the current GL
context (if valid) as if there was a conflict adding it to the display. If the
timing was unlucky, subsequent use of the GL context would be referencing an
already unreffed GL context object resulting in a critical:
g_object_ref: assertion 'object->ref_count > 0' failed
https://bugzilla.gnome.org/show_bug.cgi?id=766703
The gltestsrc element uses two shaders: color_shader and snow_shader.
Those are alternatively assigned to the SrcShader->shader pointer and
their reference was transferred to it. Only the SrcShader->shader was
unreffed (in _src_shader_deinit()) so only one shader was properly
freed, the other one was leaked.
Fixed this by giving an extra ref to SrcShader->shader and unreffing the
2 shaders in _src_smpte_free().
https://bugzilla.gnome.org/show_bug.cgi?id=766661
