Commit graph

1169 commits

Author SHA1 Message Date
Hou Qi
105d232fde v4l2bufferpool: queue back the buffer flagged LAST but empty
Some decoder drivers need to wait enough capture buffers before
starting to decode. But the dequeued buffer flag LAST but empty
has no chance to queue back to driver, which makes decode hang
after seek. So need to queue back such kind of buffer to driver.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6579>
2024-04-15 18:07:17 +00:00
Philipp Zabel
e1f5bacf8d v4l2: bufferpool: Drop writable check on output pool process
Output buffers don't have to be writable. Accepting read-only buffers
from the V4L2 buffer pool allows upstream elements to write directly
into the V4L2 buffers without triggering a CPU copy into a new buffer
from the same V4L2 buffer pool every time.

Tested with the vivid output device:

  GST_DEBUG=GST_PERFORMANCE:7 gst-launch-1.0 videotestsrc ! v4l2sink device=/dev/video5

With this change, gst_v4l2_buffer_pool_dqbuf() must be allowed to not
resize read-only memories of output buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6572>
2024-04-15 17:11:00 +00:00
Philippe Normand
111cc8d796 vpxdec: Include vpx error details in errors and warnings
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6626>
2024-04-13 10:57:43 +01:00
Philippe Normand
bd83046193 vp9enc: Include vpx error details in errors and warnings
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6626>
2024-04-13 10:56:29 +01:00
Philippe Normand
73ce4fd770 vpxenc: Rename GST_VPX_WARN to GST_VPX_ENC_WARN
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6626>
2024-04-13 10:55:55 +01:00
Qian Hu (胡骞)
cd95d02032 qtdemux: fix wrong full_range offset when parsing colr box
use colr_data[18] >> 7 to get full range information, instead
of colr_data[17] >> 7

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6616>
2024-04-12 16:59:33 +08:00
William Wedler
942415dce0 fix: qml6glsink: Notify that the returned QSGNode node has changes
Sets the QSGNode::DirtyMaterial bit when a new buffer is used for the material's texture

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3469
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6609>
2024-04-11 14:21:04 +00:00
Jochen Henneberg
687b3a2027 qt6: Let plugin documentation show up
* Added qml6 to plugin cache
* Added 'since' markers
* Moved plugin to plugins-good where it really is
* Fixed section comments

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6603>
2024-04-11 12:38:59 +00:00
Jochen Henneberg
8b87d7bcf7 qt: Let plugin documentation show up
* Enabled cc file parsing from hotdoc
* Moved package to plugins-good where it really is
* Fixed section comments

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6603>
2024-04-11 12:38:59 +00:00
Jochen Henneberg
fee46dee28 qt6: Added support for NV12 input format to qml6glsink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6582>
2024-04-10 13:45:26 +02:00
Jochen Henneberg
7065d540ee qt: Added support for NV12 input format to qmlglsink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6582>
2024-04-10 13:45:26 +02:00
Jimmy Ohn
a6c8c6f866 pulsedeviceprovider: Add is_default_device_name function and missing lock
Add is_default_device_name function to simplify compare device type
name and fix the missing lock when accessing default_sink_name and
default_source_name.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6558>
2024-04-10 08:43:56 +00:00
Arun Raghavan
82b10e57b0 pulsesink: Re-enable emission of stream status messages
This was disabled almost 10 years ago because we were missing libpulse API to
avoid a deadlock. That was fixed quite a long time ago, so let's enable this
again. The defer counter becomes an atomic, as we no longer have a threaded
mainloop lock protecting it.

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3444
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6500>
2024-04-09 15:50:04 +00:00
Philippe Normand
8d3e7689e1 vpxenc: Include vpx error details in errors and warnings
The vpx_codec_t err_detail string usually provides additional context about the
error, so include it in GStreamer warnings and errors, when it's not NULL.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6573>
2024-04-09 14:40:21 +00:00
Jochen Henneberg
6e33a5da14 qt6: Fixes for dummy texture
* 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>
2024-04-08 20:05:10 +02:00
Jochen Henneberg
87dc22b053 qt: Fixup for dummy textures
* Initialize dummy texture Ids
* Ensure YUV->RGB matrix set for dummy textures

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6578>
2024-04-08 20:05:09 +02:00
Sebastian Dröge
0596871b98 rtpbin: Don't re-use a variable for a completely different purpose temporarily
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>
2024-04-08 10:29:54 +00:00
Sebastian Dröge
11ce209ea0 rtpbin: Convert clock-base to extended RTP timestamp correctly
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>
2024-04-08 10:29:54 +00:00
Sebastian Dröge
0c34c85f7a rtpjitterbuffer: Use an extended RTP timestamp for the clock-base
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>
2024-04-08 10:29:54 +00:00
Sebastian Dröge
4a4eb56fc2 rtspsrc: Optionally timestamp RTP packets with their receive times in TCP/HTTP mode
Until https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6509
this was accidentally done inside rtpjitterbuffer for many years, and
doing so potentially solves problems on some streams while introducing
problems on others.

Make this configurable on rtspsrc and default to not set timestamps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6529>
2024-04-08 08:34:38 +00:00
Jan Schmidt
832a517965 rtpjitterbuffer: Don't use estimated_dts to do default skew adjustment
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>
2024-04-07 12:24:58 +00:00
Sebastian Dröge
ee566b8960 flac: Add wrap file and add fallback for it to the flac plugin
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6553>
2024-04-07 11:12:51 +00:00
Tim Blechmann
1c9fe19b23 v4l2: enforce a pixel aspect ratio of 1/1 if no data are available
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6242>
2024-04-07 10:14:18 +00:00
Philipp Zabel
6f9872cb56 v4l2: allocator: Fix unref log/trace on memory release
Use gst_object_unref() instead of g_object_unref() in
gst_v4l2_allocator_release(), so refcounting log and
tracer get to know about this unref.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6551>
2024-04-06 11:44:27 +00:00
Elliot Chen
e4ee4ca716 v4l2: fix error in calculating padding bottom for tile format
This is a regression while porting to arbitrary tile dimensions
introduced in !3424.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6480>
2024-04-05 13:28:47 +00:00
Elizabeth Figura
c308f013a7 atdec: Handle channel counts greater than 2
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6157>
2024-04-05 06:54:24 +00:00
Elizabeth Figura
277d6ddf22 atdec: Use gst_audio_decoder_set_output_caps() directly
The code currently sets the same caps in two different ways, and neither of them correctly handle the channel mask.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6157>
2024-04-05 06:54:24 +00:00
Sebastian Dröge
16f69acf30 wavpackparse: Use an unsigned integer for the block size calculations
It's never negative.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6498>
2024-04-04 15:10:02 +00:00
Sebastian Dröge
eefb7c1638 wavpackparse: Fix potential integer overflow on ID_ODD_SIZE blocks
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6498>
2024-04-04 15:10:02 +00:00
Sebastian Dröge
6402978a48 wavpackparse: Explicitly handle ID_WVX_NEW_BITSTREAM
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6498>
2024-04-04 15:10:02 +00:00
Robert Guziolowski
52638c1b22 qml6glsink: fix destruction of underlying texture
One should not directly delete the QRhiTexture instance.
Instead it should be marked as to be deleted once QRhi::endFrame()
is called (see: https://doc.qt.io/qt-6/qrhiresource.html#deleteLater )

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3443
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6467>
2024-04-02 11:55:16 +11:00
Tim-Philipp Müller
ef5b8dc96a tests: rtpred: fix out-of-bound writes
Don't write more data to the buffer than we allocated
space for.

Fixes #3312

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6474>
2024-03-28 19:51:47 +00:00
Haihua Hu
37e3a38ba9 v4l2src: need maintain the caps order in caps compare when fixate
if the calculated "distance" of caps A and B from the preference are
equal, need to keep the original order instead of swap them

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6451>
2024-03-28 12:53:01 +00:00
Jan Schmidt
351936aeac rtpmp4adepay: Set duration on outgoing buffers
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>
2024-03-27 10:53:38 +00:00
Sebastian Dröge
e0dfb3d974 rtphdrext-ntp: Fix typo of the RFC number in the element metadata
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3417

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6439>
2024-03-26 14:37:47 +02:00
Hou Qi
024d3ab051 v4l2: Also set max_width/max_height if enum framesize fail
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>
2024-03-22 16:02:51 +00:00
Edward Hervey
5280f0b733 adaptivedemux2: Add libsoup tracing debug
Provides more information for debugging

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6409>
2024-03-20 09:48:12 +00:00
Edward Hervey
3d500636a9 adaptivedemux2: Don't use g_str_equal on potentially NULL strings
It is only meant to be used as a callback. The fallback macro uses strcmp which
doesn't handle NULL strings gracefully. Instead use g_strcmp0

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6392>
2024-03-19 13:25:41 +00:00
Edward Hervey
ab11c20d59 hlsdemux2: Avoid NULL pointer usage
The pending/current variant are both NULL when the demuxer is resetted.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6392>
2024-03-19 13:25:41 +00:00
Edward Hervey
46bb0bfa57 adaptivedemux2: Handle context going away
This issue can happen when the scheduler loop was stopped (and context went
away). We no longer want to push/pop main context threads.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6392>
2024-03-19 13:25:41 +00:00
Edward Hervey
8438c3f567 hlsdemux2: Improve detection of playlist updates
In the case we are not updating an existing playlist, we only want to reset the
download error count if the URI we are downloading was not the previous one we
were trying to load

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6392>
2024-03-19 13:25:41 +00:00
Alexander Slobodeniuk
650534c940 rtspsrc: remove 'deprecated' flag from the 'push-backchannel-sample' signal
It seems that it was added by accident when copying from push-backchannel-buffer

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6355>
2024-03-13 19:32:46 +00:00
Sebastian Dröge
38011a01dc mpg123audiodec: Correctly handle the case of clipping all decoded samples
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3365

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6318>
2024-03-13 12:48:36 +00:00
Piotr Brzeziński
e9802f5f41 macos: Add Apple AAC encoder (atenc)
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>
2024-03-12 19:50:06 +00:00
Piotr Brzeziński
d3fba31da0 macos: Move atdec from applemedia (-bad) to osxaudio (-good)
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>
2024-03-12 09:55:10 +00:00
Piotr Brzeziński
9c084faa75 qtdemux: Fix wrapping temporary memory in buffers
That memory can disappear at any moment, doesn't cost much to just copy those few bytes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6320>
2024-03-11 18:18:01 +00:00
Nirbheek Chauhan
3bed35c342 rtspsrc: Don't invoke close when stopping if we've started cleanup
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>
2024-03-11 09:15:50 +00:00
François Laignel
7d5bb1ea7a webrtc: add all SSRC attributes getting CAPS for a PT
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>
2024-03-08 10:28:15 +00:00
Michael Tretter
5b3082257e meson: Fix description in qt options
The qt-x11 description contains a copy/paste error from the qt-wayland option.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6292>
2024-03-08 02:14:11 +00:00
Mathieu Duponchelle
519546aea3 rtpgstpay: flush on EOS
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5173>
2024-03-07 14:02:33 +00:00