* RED_OR_ALPHA8 will map value to alpha for OpenGL, use R8 to avoid
2nd shader
* Determine texel size for proper texture memory preparation
* QByteArray::fromRawData() does shallow copy and thus leads to use of
corrupted memory
* Make sure RGBA dummy texture is fully opaque
* QRhiTexture::create() must be called to allocate texture resources
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6578>
During RTP-Info synchronization, clock_base was temporarily switched
from the actual clock-base to the base RTP time and then back some lines
later.
Instead directly work with the base RTP time. The comment about using a
signed variable for convenience doesn't make any sense because all
calculations done with the value are unsigned.
Similarly, rtp_clock_base was overridden with the rtp_delta when
calculating it, which was fine because it is not used anymore
afterwards. Instead, introduce a new variable `rtp_delta` to make this
calculation clearer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6536>
It's not in the same period as the current RTP base time but always in
the very first period. This avoids using it again at a much later time.
The code in question is only triggered with rtcp-sync=rtp-info.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6536>
It is compared to other extended RTP timestamps all over rtpjitterbuffer
and since 4df3da3bab the initial extended RTP timestamp is not equal
anymore to the plain RTP time.
Continue passing a non-extended RTP timestamp via the `sync` signal for
backwards compatibility. It will always be a timestamp inside the first
extended timestamp period anyway.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6536>
When the buffer DTS is estimated based on arrival time at the
jitterbuffer (rather than provided on the incoming buffer itself),
it shouldn't be used for skew adjustment. The typical case is
packets being deinterleaved from a tunnelled TCP/HTTP RTSP stream,
and the arrival times at the jitter buffer are not well enough
correlated to usefully do skew adjustments.
This restores the original intended behaviour for the 'estimated dts'
path, that was broken years ago during other jitterbuffer refactoring.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6509>
If we can calculate timestamps for buffers, then set the duration
on outgoing buffers based on the number of samples depayloaded.
This can fix the muxing to mp4, where otherwise the last packet
in a muxed file will have 0 duration in the mp4 file.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6447>
Some driver doesn't implement enum_framesize. The maximum supported
size can be got by trying format with a very large size. Also need
to set max_width/max_height for this case, otherwise default encoded
buffer size 256kB is too small.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6416>
Adds the `atenc` element capable of encoding AAC-LC audio, using the AudioToolbox framework.
It's able to encode up to 7.1 channel configurations.
Comes with basic knobs for rate control (bitrate for CBR, quality for VBR).
Support for more profiles (LD, HE-AAC) should be simple, but is not included here because of bugs
with parsing of the AudioSpecificConfig.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6254>
osxaudio has a few helper methods potentially useful in atdec (or future atenc), like GStreamer -> CoreAudio
channel mapping. Doesn't make sense to duplicate them in applemedia, and atdec is the only audio-oriented
element there anyway.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6223>
When we're doing a state change from PLAYING to NULL, first we invoke
gst_rtspsrc_loop_send_cmd_and_wait (..., CMD_CLOSE, ...) during
PAUSED_TO_READY which will schedule a TEARDOWN to happen async on the
task thread.
The task thread will call gst_rtspsrc_close(), which will send the
TEARDOWN and once it's complete, it will call gst_rtspsrc_cleanup()
without taking any locks, which frees src->streams.
At the same time however, the state change in the app thread will
progress further and in READY_TO_NULL it will call gst_rtspsrc_stop()
which calls gst_rtspsrc_close() a second time, which accesses
src->streams (without a lock again), which leads to simultaneous
access of src->streams, and a segfault.
So the state change and the cleanup are racing, but they almost always
complete sequentially. Either the cleanup sets src->streams to NULL or
_stop() completes first. Very rarely, _stop() can start while
src->streams is being freed in a for loop. That causes the segfault.
This is unlocked access is unfixable with more locking, it just leads
to deadlocks. This pattern has been observed in rtspsrc a lot: state
changes and cleanup in the element are unfixably racy, and that
foundational issue is being addressed separately via a rewrite.
The bandage fix here is to prevent gst_rtspsrc_stop() from accessing
src->streams after it has already been freed by setting src->state to
INVALID.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6302>
The transport stream only returned the CAPS for the first matching PT entry
from the `ptmap`. Other SSRC with the same PT where not included. For a stream
which bundled multiple audio streams for instance, only the first SSRC was
knowed to the SSRC demux and downstream elements.
This commit adds all the `ssrc-` attributes from the matching PT entries.
The RTP jitter buffer can now find the CNAME corresponding its SSRC even if it
was not the first to be registered for a particular PT.
The RTP PT demux removes `ssrc-*` attributes cooresponding to other SSRCs
before pushing SSRC specific CAPS to downstream elements.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6119>
And also re-timestamp them with the current buffer's PTS.
Not doing so keeps the timestamps of event packets as
GST_CLOCK_TIME_NONE or the timestamp of the previous buffer, both of
which are bogus.
Making sure that (especially) the first packet has a valid timestamp
allows putting e.g. the NTP timestamp RTP header extension on it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5173>
Parse the speed and scale in the server's response
*before* the range, so that the range start/stop
are swapped (or not swapped) correctly based
on the server's actual chosen values. Otherwise,
the old rate from the segment is used - what the
last seek asked for, but not necessarily what
the server chooses.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6248>
If input height and parsed one are identical, do not consider it as interlaced
Fixing below pipeline:
gst-launch-1.0 videotestsrc ! video/x-raw,format=I420,width=640,height=10 \
! jpegenc ! jpegparse ! jpegdec ! videoconvert ! autovideosink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6181>
After a flushing seek, rtspsrc doesn't reset the last_ret value for
streams, so might immediately shut down again when it resumes pushing
buffers to pads due to a cached `GST_FLOW_FLUSHING` result
Prevent a stored flushing value from immediately stopping
playback again by resetting pad flows before (re)starting
playback.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6137>
Because this depayloader may build several output buffers within one
process run we push them all into a GstBufferList and push them out at
once to make sure that each buffer gets notified about each header
extension.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
Because this depayloader may build several output buffers within one
process run we push them all into a GstBufferList and push them out at
once to make sure that each buffer gets notified about each header
extension.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
Because this depayloader may build several output buffers within one
process run we push them all into a GstBufferList and push them out at
once to make sure that each buffer gets notified about each header
extension.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
Because this depayloader may build several output buffers within one
process run we push them all into a GstBufferList and push them out at
once to make sure that each buffer gets notified about each header
extension.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
Because this depayloader may build several output buffers within one process
run we push them all into a GstBufferList and push them out at once to
make sure that each buffer gets notified about each header extension.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378>
In rtpbin we already systematically check for all property names
except latency, correct that.
In webrtcbin we need to check before trying to use the do-retransmission
property.
This is useful for the case where an element like identity gets passed
to rtpbin's request-jitterbuffer property, when the application wants
to use webrtcbin in an SFU situation, with no reordering and no added
latency
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6112>
Today when using the `splitmuxsrc` on a collection of files named as:
```
item0.mkv
item1.mkv
item2.mkv
[...]
item10.mkv
item11.mkv
[...]
```
You will get a continuous stream made in the order of:
```
item0.mkv -> item1.mkv -> item10.mkv -> item11.mkv -> [...]
```
You can fix this by having smarter names of the items:
```
item000.mkv
item001.mkv
item002.mkv
[...]
item010.mkv
item011.mkv
[...]
```
Will get you:
```
item000.mkv -> item001.mkv -> item003.mkv -> item004.mkv -> [...]
```
But, we could also "fix" the former case by using natural ordering when
comparing the files in gstsplitutils.c.
Fixes#2523
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4491>
When level value is greater than 127, it was being clamped but this clamped
value was not the one being actually used. For level values greater than 127
this resulted in an incorrect value being used. As an example, a level value
of 187, after and'ed with 0x7F, it would result in 0x3B being reported as the
level value.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5893>
When reopening a v4l2 device, the v4l2object->poll will include some old fds,
which was assigned to this device before. If the pipeline opens multiple v4l2
devices, the old fd may been assigned to other v4l2 devices when reopening
devices.
This will cause the timing of the pipeline become confusing when polling devices,
leading functional abnormalities.
Therefore, when closing v4l2object, remove the old fds in poll to ensure that the
pipeline timing is normal.
Signed-off-by: Chao Guo <chao.guo@nxp.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5820>
Apparently external-oes is not supported by the plugin as texture target,
while DMABuf uploading prefers it because it's zero copy.
This patch enables DMABuf uploading and rendering by using either 2D or
rectangle texture targets.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5795>
This simplifies the way it picks the closest caps to preference and take into
consideration the framerate to avoid picking high resolution at 5fps or so.
Simply calculate a "distance" of caps A and B from the preference and put
closest first, sorting by framerate first.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5777>
The `GstFlowCombiner` is responsible for tracking the flow of each
stream and handle the overal flow return value. Without that, we
can end up with the following scenario:
- Audio+video stream
- Only the video stream is linked downstream
- The audio stream goes EOS, video doesn't yet
-> We update the Flow in the combiner with OK as all streams are not EOS
- Video goes EOS because downstream returned EOS
-> `qtdemux` returns `FLOW_OK` forever because the unlinked audio pad
has `last_flowret==FLOW_OK`
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5724>
The tags and caps were leaked for unknown streams, I'm not sure they'd be valid
in that case, but better safe than sorry.
The tags ownership is transfered when calling `gst_adaptive_demux_track_new()`
so unreffing those afterwards was a mistake.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5714>
If this property is enabled then the jitterbuffer will do the normal PTS
calculations according to the configured mode instead of making use of
the RFC7273 media clock.
The timestamp calculated from the RFC7273 media clock will only be
stored in the reference timestamp meta, if addition of that meta is enabled.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5512>
When this property is used, it is assumed that the system clock is
synced close enough to the media clock used by an RFC7273 stream.
As long as both clocks are at most a few seconds from each other this
will give the correct results and avoids having to create an actual
network clock that has to sync first.
If the system clock is actually synchronized to the media clock then
everything will behave exactly the same, otherwise the reference
timestamp meta will be correct but the buffer timestamps will be off by
the difference between the two clocks.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5512>
Do more checks for clock equality than just checking pointers. The same
NTP/PTP clock might be used as pipeline clock but a new instance, so
instead also check what clock they are synced to.
Also handling setting / resetting of the media clock and pipeline clock
correctly by resetting the media clock's state accordingly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5512>
Because we treat raw audio chunks/samples as keyframes, they were interfering
with seek time adjustment.
Became apparent when the accompanying video stream was I-frame only,
for example ProRes.
Since raw audio streams can be seeked freely, it's fine to just ignore them here,
giving priority to the real keyframes in the video stream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4946>
In snapshot mode pngenc should output exactly one frame
and then return FLOW_EOS to upstream. If upstream sends
more input frames before shutting down, it should keep
returning FLOW_EOS but not output any more encoded frames.
After a flushing seek it should output frames again though.
Fixes#3069.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5546>
The default 2MB ENCODED_BUFFER_SIZE can't support some 4K video playback. We now
detect the driver reported maximum resolution and choose an appropriate
default bitstream size accordingly. For 4K video these results in around 4MB
buffer instead of 2MB.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4549>
After rendering a QML scene the qml6glsrc element copies the contents of
the scene to a GStreamer buffer. This happens on the Qt render thread.
Then it attaches a sync point to the destination buffer. This sync point
must be awaited by other threads which use the buffer later on. The
current implementation relies on the downstream elements to wait for the
sync point. However, there are situation where this does not work. The
GstBaseTransform e.g. copies the buffer metadata (which overwrites the
sync point without waiting for it) *before* waiting for the sync point.
This commit waits for the sync point inside the qml6glsrc element before
sending it downstream. The wait command is issued on the streaming
thread with the pipeline OpenGL context, i.e. it will synchronize with
the GStreamer OpenGL thread.
This is a port of the original fix for the qmlglsrc element.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5519>
After rendering a QML scene the qmlglsrc element copies the contents of
the scene to a GStreamer buffer. This happens on the Qt render thread.
Then it attaches a sync point to the destination buffer. This sync point
must be awaited by other threads which use the buffer later on. The
current implementation relies on the downstream elements to wait for the
sync point. However, there are situation where this does not work. The
GstBaseTransform e.g. copies the buffer metadata (which overwrites the
sync point without waiting for it) *before* waiting for the sync point.
This commit waits for the sync point inside the qmlglsrc element before
sending it downstream. The wait command is issued on the streaming
thread with the pipeline OpenGL context, i.e. it will synchronize with
the GStreamer OpenGL thread.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5506>
With one regular image file path provided (without %05d),
the element was stuck in a dead loop counting the frames:
gst_image_sequence_src_count_frames
This allows to display any image file out of the element
for a given number of buffers.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5471>
If the v4l2videoenc receives an QUERY_ALLOCATION, it must not propose a
currently used pool, because it cannot be sure that the allocation query came
from exactly the same upstream element. The QUERY_ALLOCATION will not contain
the internal OUTPUT pool.
The upstream element (the basesrc) detects that the newly proposed pool differs
from the old pool. It deactivates the old pool and switches to the new pool.
If there was a format change, a new OUTPUT buffer pool will be allocated in
gst_v4l2_object_set_format_full() and the CAPTURE task will be stopped to switch
the format. If there hasn't been a format change,
gst_v4l2_object_set_format_full() will not be called. The old pool will be kept
and reused.
Without a format change, the processing task continues running.
This leads to the situation that the processing task is running, but the OUTPUT
buffer pool (the old pool) is deactivated. Therefore, the encoder is not able to
get buffers from the OUTPUT pool and encoding cannot continue.
This situation can be triggered by sending a RECONFIGURE event without a format
change.
Resolve this situation by ensuring that the OUTPUT buffer pool is always
activated when frames arrive at the encoder.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4235>
There is a CAPTURE pool in the same function. While the CAPTURE pool is called
cpool, using pool for the OUTPUT pool is confusing.
Using opool for the OUTPUT pool makes it more obvious, which pool is used.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4235>
We were already converting the pad last timestamp to running time but
not the segment position.
This segment position is used by gst_aggregator_simple_get_next_time()
to compute the waiting time when aggregating.
Those waiting times were wrong in my live pipeline using the system
clock, resulting in the aggregator to never wait at all.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5460>
While adding arbitrary tile support, a round up operation was badly
converter. This caused the Y component of the stride to be 0. This
eventually lead to a crash in glupoad preceded by the following
assertion.
gst_gl_buffer_allocation_params_new: assertion 'alloc_size > 0' failed
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5458>
There is a race condition where transfer has not been submitted yet while the
request is cancelled which leads to the transfer state going back to
`DOWNLOAD_REQUEST_STATE_OPEN` and the user of the request to get signalled about
its completion (and the task actually happening after it was cancelled) leading
to assertions and misbehaviours.
To ensure that this race can't happen, we start differentiating between the
UNSENT and CANCELLED states as in the normal case, when entering `submit_request`
the state is UNSENT and at that point we need to know that it is not because
the request has been cancelled.
In practice this case lead to an assertion in
`gst_adaptive_demux2_stream_begin_download_uri` because in a previous call to
`gst_adaptive_demux2_stream_stop_default` we cancelled the previous request and
setup a new one while it had not been submitted yet and then got a `on_download_complete`
callback called from that previous cancelled request and then we tried to do
`download_request_set_uri` on a request that was still `in_use`, leading to
something like:
```
#0: 0x0000000186655ec8 g_assert (request->in_use == FALSE)assert.c:0
#1: 0x00000001127236b8 libgstadaptivedemux2.dylib`download_request_set_uri(request=0x000060000017cc00, uri="https://XXX/chunk-stream1-00002.webm", range_start=0, range_end=-1) at downloadrequest.c:361
#2: 0x000000011271cee8 libgstadaptivedemux2.dylib`gst_adaptive_demux2_stream_begin_download_uri(stream=0x00000001330f1800, uri="https://XXX/chunk-stream1-00002.webm", start=0, end=-1) at gstadaptivedemux-stream.c:1447
#3: 0x0000000112719898 libgstadaptivedemux2.dylib`gst_adaptive_demux2_stream_load_a_fragment [inlined] gst_adaptive_demux2_stream_download_fragment(stream=0x00000001330f1800) at gstadaptivedemux-stream.c:0
#4: 0x00000001127197f8 libgstadaptivedemux2.dylib`gst_adaptive_demux2_stream_load_a_fragment(stream=0x00000001330f1800) at gstadaptivedemux-stream.c:1969
#5: 0x000000011271c2a4 libgstadaptivedemux2.dylib`gst_adaptive_demux2_stream_next_download(stream=0x00000001330f1800) at gstadaptivedemux-stream.c:2112
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5435>
The counter was using a signed 8 bit integer, which was overflowing
after 127 entries. That was then passed as an unsigned 32 bit integer to
libflac, which caused it to be converted to a huge unsigned number.
That then caused an invalid memory access inside libflac.
As a bonus, signed integer overflow is undefined behaviour.
Instead, use an unsigned 8 bit integer. Once this overflows the existing
code already catches it and stops adding the cue. While FLAC__metadata_object_cuesheet_insert_track()
takes an unsigned 32 bit integer for the track number, FLAC__StreamMetadata_CueSheet_Track is
limiting it to an unsigned 8 bit integer.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2921
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5420>
Found that osxaudiosink could not be added standalone in gst-full build
using
-Dgst-full-elements=osxaudio:osxaudiosink because element registration
was
done at the plugin level. Now src/sink elements and deviceprovider have
their
individual registration.
Copied/adapted from the alsa plugin.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5419>
scanlines->m1 = same line of the previous field
scanlines->t0 = line above of the current field
scanlines->b0 = line below of the current field
scanlines->mp = same line of the next field
Deinterlacing a field weaved frame:
When deinterlacing the top field, the next bottom field is available
(part of the same frame). but when deinterlacing the bottom field,
the next top field (part of the next frame) is not available and
scanlines->mp equals NULL.
In this case it's better to use greedy algorithm using the prevous field
(twice) rather then linear interpolation of the current field.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5331>
If we end up with GST_CLOCK_TIME_NONE as running time for an RTP packet
then this can't be used for bitrate estimation, and also not for
constructing the next RTCP SR. Both would end up with completely wrong
values, and an RTCP SR with wrong values can easily break
synchronization in receivers.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5329>
The timestamp offset can be negative, and it can be a bigger negative
number than the latency introduced by the rtpjitterbuffer so the overall
timeout offset can be negative.
Using the negative offset for calculating how many packets can still
arrive in time when encountering a lost packet in an equidistant stream
would then overflow and instead of considering fewer packets lost a lot
more packets are considered lost.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5296>
gst_video_info_set_interlaced_format() can return an error if the
width/height causes integer overflow. Handle this case, so that we can
fail cleanly. This has been experienced while testing an in-progress
driver.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5286>
Some drivers will push an buffer flagged LAST but empty. In decoder
case, this results in an "producing too many buffer" warning, even
though the result is entirely correct. Detect this case in order to
signal EOS earlier and avoid this warning.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5286>
when playing some codec such as matroska with vp9 codec,
demuxer will save information like video_mastering_display_info
and video_content_light_level in caps that decoder need,
v4l2videodecoder can use it by calling V4L2_CTRL_CLASS_COLORIMETRY
ioctl.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4403>
If decoder notify a source change event when the capture format is
changed, not the resolution changed.
then gst_v4l2_object_acquire_format will retuen false due to
unsupported format.
we need to clear the format lists in the source change flow,
and reenumerate format list
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5218>
This elements pass RTP packets along unchanged and appear as a RTP
payloader element.
This is useful, for example when using the gstreamer-rtsp-server
library, in the case where you are receiving RTP packets from a
different source and want to serve them over RTSP. Since the
gst-rtsp-server library expect the element marked as payX to be a RTP
payloader element and assumes certain properties are available.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5204>
Imported dmabuf are not being duped, so they should never be closed. Instead,
we ensure their live time by having strong reference on their original
buffer. This should fix potential flickering due to dmabuf being closed
too early.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5101>
Now that we can split GStreamer buffers over multiple v4l2 buffer, we may
endup waiting for these buffers to be processed. Avoid waiting for any of
the parts being processed. As a side effect, the pool will now try to
grow if the number of buffers is not sufficient, and will fail
otherwise.
This fixes a hang if the very first frame did not fit. In this case, the
driver will retrain that buffer until the capture is setup, but
GStreamer won't setup the capture until process() function have
returned.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5100
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5143>
The hack enforcing strictly increasing timestamps was, according to the
code comments, because librtmp was confused with backwards timestamps.
rtmp2sink is not using librtmp as rtmpsink did, so this is no longer
required.
Also changing the timestamps is causing audio glitches when streaming to
Youtube.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5212>
These 10bit formats are identical to NV12_16L32S, but 64bytes of data is being
prefixed with 16bytes data with four pixels of lower 2bits per byte. For
MT2110T, the lower two bits set so each bytes contains a column of 4 pixels,
also describe as tiled lower 2 bits. MT2110T has been chosen as a name to match
the vendor chosen name. This format is unlikely to exist for other vendors.
For MT2110R, the 2 low bits are in raster order.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3444>
When advancing the ringbuffer, store the processed CoreAudio sample
time, then interpolate the clock in the _get_delay() calls to smooth
the clock. CoreAudio's "latency" report is always a constant and
otherwise leads to the clock generating a latency-time staircase.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5140>
Set the BufferFrame size in CoreAudio so it will deliver data
in ringbuffer segment units when recording. Otherwise, CoreAudio
will provide data in whatever granularity it wants, with no
relationship to the requested latency-time.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5140>
Issue was that Qt was caching multiple different shadersbased on a static
variable location. This static variable location was the same no matter
the input video format and so if an item had an input video format of
RGB and another of RGBA, they would eventually end up using the same
GL shader leading to incorrect colours.
Fix by providing different static variable locations for each of the
different shaders that will be produced.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5145>
When we fill a bitstream buffer the buffer might be too small to hold
the entire frame. Only resize to the filled size, preventing the
following assertion to happen.
gst_buffer_resize_range: assertion 'bufmax >= bufoffs + offset + size' failed
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5100>
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2771
This EOS branch exists so that if a seek with a stop is made, qtdemux
stops accepting bytes from the sink after the entire requested playback
range is demuxed, as otherwise we could keep download content that is
not being used.
This patch fixes two flaws that were present in that EOS check:
1) A comparison was made between track time and movie time without conversion.
This made the check trigger early in files with edit lists. This patch fixes
this by converting the track PTS to movie PTS (stream time) for the check.
2) To avoid sending a EOS prematurely when the segment stop is within a GOP and
B-frames are present, the check for EOS should only be done for keyframes. I
gather this was already the intention with the existing code, but because it
used `stream->on_keyframe` instead of the local variable `keyframe` the old
code was checking if the *previous* frame was a keyframe.
It's interesting to note that these two flaws in the old code mask each other
in most cases: the track PTS will have reached the movie end PTS, but EOS would
only be sent if the previous frame was a keyframe. A simple case where they
wouldn't mask each other, reproducing the bug, is a sequence of 3 frame GOPs
with structure I-B-P.
The following validateflow tests have been added to future-proof the
fix:
* validate.test.mp4.qtdemux_ibpibp_non_frag_pull.default
* validate.test.mp4.qtdemux_ibpibp_non_frag_push.default
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5021>
We were checking if the tag list is writable, but it may actually be
shared through the same event (tee upstream or multiple consumers).
Fix a bug where multiple branches have a videoflip element checking the
taglist. The first one was changing the orientation back to rotate-0
which was resetting the other instances.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5097>
This check fixes a critical warning that can happen when a pointer motion
happens and the video doesn't have its width/height information available.
GStreamer-Video-CRITICAL **: gst_video_center_rect: assertion 'src->h != 0' failed
#0 g_logv (log_domain=0x7ffff705e176 "GStreamer-Video", log_level=G_LOG_LEVEL_CRITICAL, format=<optimized out>, args=<optimized out>) at ../../../../Projects/jhbuild/glib/glib/gmessages.c:1422
#1 0x00007ffff7e1a81d in g_log (log_domain=<optimized out>, log_level=log_level@entry=G_LOG_LEVEL_CRITICAL, format=format@entry=0x7ffff7e77a9d "%s: assertion '%s' failed") at ../../../../Projects/jhbuild/glib/glib/gmessages.c:1460
#2 0x00007ffff7e1b749 in g_return_if_fail_warning (log_domain=<optimized out>, pretty_function=<optimized out>, expression=<optimized out>) at ../../../../Projects/jhbuild/glib/glib/gmessages.c:2930
#3 0x00007ffff701d90b in gst_video_sink_center_rect (src=..., dst=..., result=result@entry=0x7fffffffc6d0, scaling=scaling@entry=1) at ../../../../Projects/jhbuild/gstreamer/subprojects/gst-plugins-base/gst-libs/gst/video/gstvideosink.c:105
#4 0x00007fffe5652dbb in _fit_stream_to_allocated_size (result=0x7fffffffc6d0, allocation=0x7fffffffc6c0, base_widget=0x9396f0) at ../../../../Projects/jhbuild/gstreamer/subprojects/gst-plugins-good/ext/gtk/gtkgstbasewidget.c:326
#5 gtk_gst_base_widget_display_size_to_stream_size (base_widget=base_widget@entry=0x9396f0, x=1207.7109375, y=811.84765625, stream_x=stream_x@entry=0x7fffffffc720, stream_y=stream_y@entry=0x7fffffffc728) at ../../../../Projects/jhbuild/gstreamer/subprojects/gst-plugins-good/ext/gtk/gtkgstbasewidget.c:344
#6 0x00007fffe5651a4b in gst_gtk_base_sink_navigation_send_event (navigation=0x5ff990, event=0x178a730) at ../../../../Projects/jhbuild/gstreamer/subprojects/gst-plugins-good/ext/gtk/gstgtkbasesink.c:340
#7 0x00007fffe5652432 in gtk_gst_base_widget_motion_event (widget=<optimized out>, event=event@entry=0x1f14b60) at ../../../../Projects/jhbuild/gstreamer/subprojects/gst-plugins-good/ext/gtk/gtkgstbasewidget.c:404
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5049>
The libpsl subproject wasn't building successfully and CI didn't
notice because:
1. The plugin wasn't explicitly enabled
2. Even when the plugin is explicitly enabled, the dep is not required
at build time when not building a static plugin
So fix all of these issues.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5038>
The videoencoder base class uses getcaps() to ask a subclass for the caps in its
sink_query_default() implementation.
Replace the custom handling of the QUERY_CAPS in the v4l2videoenc with an
implementation of getcaps() that returns the caps that are supported by the
v4l2videoenc to return these caps in the query.
This getcaps() implementation also calls the provided proxy_getcaps(), which
sends a caps query to downstream. This fixes the v4l2videoenc element to respect
limits of downstream elements in a sink query.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5034>
The videoencoder base class always uses the negotiated allocator for allocating
coded buffers and ignores the negotiated buffer pool. Therefore, the
v4l2videoenc always has to copy buffers from the pool into the allocated
output buffers.
This breaks downstream elements that want to import the CAPTURE buffers of the
v4l2videoenc, since the v4l2videoenc copies the exported CAPTURE buffers and
sends the copies downstream.
Always use the CAPTURE buffer pool for acquiring CAPTURE buffers instead of
allocating the buffers in the base class.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4230>
If the capture pool is already active, like when handling gaps at the
start of a stream, do not setup the decoder to wait for src_ch event.
Otherwise the decoder will endup waiting for that at the wrong moment
and exit the decoding thread unexpectedly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4590>
Fix this pipeline where the tag list is not writable:
gst-launch-1.0 videotestsrc ! taginject tags="image-orientation=rotate-90" ! videoflip video-direction=auto \
! autovideosink
GStreamer-CRITICAL **: 12:34:36.310: gst_tag_list_add: assertion 'gst_tag_list_is_writable (list)' failed
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4987>
This fixes a build error if Qt was build without accessibility support:
../../../../../gstreamer/subprojects/gst-plugins-good/ext/qt/qtglrenderer.cc:
In member function 'bool GstQuickRenderer::init(GstGLContext*, GError**)':
../../../../../gstreamer/subprojects/gst-plugins-good/ext/qt/qtglrenderer.cc:341:13:
error: 'QCoreApplication' was not declared in this scope; did you mean 'QApplication'?
../../../../../gstreamer/subprojects/gst-plugins-good/ext/qt/qtglrenderer.cc:341:31:
error: 'app' was not declared in this scope
../../../../../gstreamer/subprojects/gst-plugins-good/ext/qt/qtglrenderer.cc:341:37:
error: 'QCoreApplication' is not a class, namespace, or enumeration
[...]
../../../../../gstreamer/subprojects/gst-plugins-good/ext/qt/qtglrenderer.cc:458:5:
error: 'QEventLoop' was not declared in this scope; did you mean 'QEvent'?
../../../../../gstreamer/subprojects/gst-plugins-good/ext/qt/qtglrenderer.cc:459:9:
error: 'loop' was not declared in this scope
If accessibility is enabled, the includes for QCoreApplication and QEventLoop
are indirectly pulled via QWidget.
Add the required headers as documented in [1] and [2].
[1] https://doc.qt.io/qt-5/qcoreapplication.html
[2] https://doc.qt.io/qt-5/qeventloop.html
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4815>
Previously, we would create a new GstMemory per write operation
and then append them to the GstBuffer. This would cause a reallocation
every 16 Memories which is an issue since the png encoder will usually
do write in a pattern of 4, 8 and 8k bytes repeating until the frame
is done.
Instead allocate a single GstMemory and keep writting it into it
with a manual index. Much like the jpegenc does.
Doing some basic testing With a testsrc snow pattern at 4k and 8k
the same pipeline would take ~3.30s to encode a 4k frame and ~23s
for an 8k. At 4k 0.70s/33% is taken by memory allocations, while at
8k its ~10.5s/45%.
With this patch, at 4k the pipeline takes ~2.40s and at 8k only 9.60s
making this 28% and 58% faster accordingly on my laptop, and
allocation runtime is dropped to subsecond times.
Here's the test pipeline used, increase num-buffers in image freeze
to gather more samples.
```
gst-launch-1.0 videotestsrc num-buffers=1 pattern=snow ! imagefreeze num-buffers=1 ! \
video/x-raw,width=7680,height=4320 ! pngenc ! fakesink
```
Close#2717
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4944>
The aim of this example is to show how to make use of the accept-certificate
signal from a GTK GUI, and prompt user in case of invalid certificate.
There are two subtleties to be aware of:
1. the signal is emitted from the GStreamer streaming thread, therefore the
caller can't modify the GUI straight away, instead they must do it from the
main thread (eg. by using g_idle_add())
2. in case of a redirection, then a TLS failure, the caller won't know
about the redirection. Actually, it's possible to be notified of the
redirection by watching "message:element" and inspecting http-headers,
but even in that case, the signal will be received *after* the signal
"accept-certificate" (even though the redirection happened *before*).
This second point is tricky. It's not uncommon to have servers that redirect
http requests to https. So errors of the type "HTTP -> HTTPS -> TLS error"
happen, and if the caller doesn't care about redirection, they might prompt
users with a message such as "TLS error for URL http://...", which wouldn't make
much sense.
This example shows how to handle that right, by connecting to the signal
"message:element", inspecting the http-headers, and in case of redirection,
updating the TLS error dialog to indicate that the request was redirected.
Here are a few examples of streams that exhibit TLS failure (at the time of
this commit, of course):
* https://radiolive.sanjavier.es:8443/stream: unknown-ca
* https://am981.ddns.net:9005/stream.ogg: unknown-ca
* http://stream.diazol.hu:7092/zene.mp3: redir then bad-identity
* https://streaming.fabrik.fm/izwi/echocast/audio/index.m3u8: unknown-ca
(this one is a HLS stream)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4925>
With libsoup 2.x, it was possible to know when there was a TLS failure, as
libsoup provided the "special http status code" SOUP_STATUS_SSL_FAILED.
However these special codes were dropped with libsoup 3.x: now libsoup emits
the accept-certificate signal when there's a TLS failure.
This commit adds a signal "accept-certificate" to SoupHttpSrc, which is in fact
just about forwarding the signal from SoupMessage (which is, itself, forwarded
from GTlsConnection). Note that, in case of libsoup 2.x, the signal is never
emitted.
Fixes: #2379
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4925>
In the current implementation, we support for most pixel format left
and top padding by changing the offset in the video meta. Though, to
align driver bytesused to the offset, we recalculate the offset, which
removed the modification we did before.
Instead, save the plane size, and truncate the driver reported bytesused
to the expected size, which ensures that the offsets still match. This
should also fix issues were the buffer size ended up bigger then the
pool size due to driver introduced padding.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4920>
Refusing an incoming segment in < GST_MATROSKA_READ_STATE_DATA should only be
done if the incoming segment is not in GST_FORMAT_TIME.
In GST_FORMAT_TIME, we are just storing the values and returning, so we can
invert the order of the checks.
Fixes proper segment propagation in matroska/webm DASH use-cases
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3914>
Is a seek is done on stream-collection post, there are no selected streams
yet. Therefore none would be chosen to adjust the key-unit seek.
If no streams are selected, fallback to a default stream (i.e. one which has
track(s) with GST_STREAM_FLAG_SELECT).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3914>
When seeking is handled by the collection posting thread, there is a possibility
that some leftover data will be pushed by the stream thread.
Properly detect and reject those early segments (and buffers) by comparing it to
the main segment seqnum
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3914>
... otherwise streams with constant size samples defined with a single
`sample_size` for all samples in the `stsz` box fall in the category
`chunks_are_samples` in `qtdemux_stbl_init`, overriding the actual
sample count.
`FOURCC_soun` would set this automatically for `compression_id == 0xfffe`,
however `compression_id` is read from the Audio Sample Entry box at an offset
marked as "pre-defined" in some version of the spec and set to 0 both by
GStreamer and FFmpeg for opus streams.
Considering the stream `sampled` flag is set explicitely by other fourcc
variants, doing so for opus seems consistent.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4903>
The "Encapsulation of Opus in ISO Base Media File Format" [1] specifications,
§ 4.3.2 Opus Specific Box, indicates that data must be stored as big-endian.
In `build_opus_extension`, `gst_byte_writer_put*_le ()` variants were used,
causing audio streams conversion to Opus in mp4 to offset samples due to the
PreSkip field incorrect value (29ms early in our test cases).
[1] https://opus-codec.org/docs/opus_in_isobmff.html#4.3.2
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4875>
- Adding bayer 10,12,14,16 bits components with 16 bits storage. These
changes only adds capabilities. Capability format string is a complete
description of the frame and pixels layout. Only mapping LE bayer
formats as v4l2 only define LE bayer formats.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4852>
When doing a segment seek, the base offset in the new segment
would be increased by segment.position which is basically the
timestamp of the last packet. This does not include the duration
of the last packet though, so might be slightly shorter than the
actual duration of the clip or the requested segment.
Increase the base offset by the segment duration instead when
accumulating segments, which is more correct as it doesn't cut
off the last frame and makes the effective loop segment duration
consistent with the actual duration returned from a duration
query.
In case a segment stop was specified it's also possible that
some data was sent beyond the stop that's necessary for decoding
so the base offset increment should be based on that then and
not on the timestamp of the last buffer pushed out.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4604>
The muxer used a fixed value of 2 channels because the TR 102 366 spec
says they're to be ignored. However, the demuxer still trusted them,
resulting in bad caps.
Make the muxer fill in the correct channel count anyway (FFmpeg already
does) and make the demuxer ignore the value.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4739>
