Commit graph

112610 commits

Author SHA1 Message Date
Stéphane Cerveau
508a565163 matroska: demux: update stream_start_time
The stream_start_time can be less than the first detected.
In case of B-Frame based media, the first frame PTS might be
greater than the next one.

Need to keep the segment.start if a seek has been performed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1030>
2021-08-17 16:09:14 -04:00
Nicolas Dufresne
65deef0b0c mastrokademux: Remove redundant assignment
The segment.position is unconditionnaly set few lines below.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1030>
2021-08-17 16:08:33 -04:00
Devarsh Thakkar
297b1e68e2 ext: alsa: Fix fallback paths for setting buffer and period times
Below fallback paths were introduced in
9759810d82
if setting period time after buffer time failed :
1) Set period time and then buffer time if it doesn't work
2) Set only buffer time
3) Set only period time

These all were not functioning properly since they were using old
copy of snd_pcm_hw_params_t which already had some fields set
as per previous try and this was causing issues as driver was
referring to that old value while trying to set them again in
fallback paths.

So now we always use the initial copy of snd_pcm_hw_params_t
for every fallback  and same is also being done at
557c429510

Also we change the sequence to set period time earlier than
buffer time since period bytes being the smaller unit, most of the times
if underlying alsa device has a dependency then it is of period bytes
to be a multiple of some value (as per underlying DMA constraint)
and rest of the parameters like buffer bytes need to be adjusted
as per period bytes.

The same sequence is also followed in alsa-utils at
9b621eeac4

Fix 2) and 3) scenarios by returning success if the exclusive setting is passed
and not doing any further setting for buffer time or period time.

Add new fallback path of not setting any buffer time and period time
if all above fallback paths fail. The same is also being
followed at aforementioned pulseaudio commit.

In case of alsasink, remove the retry goto label, since it is not
required anymore as fallback paths take care of setting default
values if driver is not accepting any of the fallback paths.

Use separate label for exit to free params structs and return err
code. This also fixes leak in no_rate goto path in alsasink

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1212>
2021-08-17 19:52:59 +00:00
Víctor Manuel Jáquez Leal
d1cd310e42 videocrop: Fix icles tests.
Internally videcrop can call gst_video_crop_set_info() with NULL as in
caps. Then critical messages are raised when the in caps are
processed.

To fix this the in caps are checked, and if they are present, its
capsfeature is extracted, otherwise, the previous raw caps detection
remains as before.

Also the videocrop-test removes the format field in the structure
because now its always passed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1056>
2021-08-17 17:19:16 +00:00
Jakub Adam
286208576f rtp: Color Space header extension
Implements WebRTC header extension defined in
http://www.webrtc.org/experiments/rtp-hdrext/color-space.

It uses RTP header to communicate color space information and optionally
also metadata that is needed in order to properly render a high dynamic
range (HDR) video stream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/853>
2021-08-17 15:28:19 +00:00
Jakub Adam
b4a00f78bc videoencoder: pass upstream HDR information through codec state
Don't copy HDR metadata from sink pad, because its caps may not have
been set yet if GstVideoEncoder::negotiate is called from
GstVideoEncoder::set_format, as e.g. vpx encoder does.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1175>
2021-08-17 14:54:06 +00:00
Jakub Adam
b3c7b9be49 videoutils: add HDR metadata fields to GstVideoCodecState
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1175>
2021-08-17 14:54:06 +00:00
Hou Qi
0e7a485528 v4l2: Add protection when set decoder capture fps accroding to output fps
Some v4l2 drivers don't have the capacity to change framerate. There is
chance to make decoder capture fps to be 0/0 if numerator and denominator
returned by G_PARM ioctl are both 0. It causes critical warning
"passed '0' as denominator for `GstFraction'".

In order to fix this, add protection when set decoder capture fps according
to output fps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1048>
2021-08-17 13:27:28 +00:00
Nirbheek Chauhan
9c7a2df4a8 cerbero: Always fetch sources with four parallel jobs
The default number of parallel jobs is two, which is too few. We can
easily use four or more. Should speed up image builds and also
downloading of (new) sources that aren't already cached in the image.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-ci/-/merge_requests/411>
2021-08-17 16:04:42 +05:30
Per Förlin
9a216d0ffa rtspsrc: Add support to ignore x-server HEADER reply
When connecting to an RTSP server in tunnled mode (HTTP) the server
usually replies with a x-server header. This contains the address
of the intended streaming server. However some servers return an
"invalid" address. Here follows two examples when it might happen.

1. A server use Apache combined with a separate RTSP process to handle
   Https request on port 443. In this case Apache handle TLS and
   connects to the local RTSP server, which results in a local
   address 127.0.0.1 or ::1 in the x-server reply. This address is
   returned to the actual RTSP client in the x-server header.
   The client will receive this address and try to  connect to it
   and fail.

2. The client use a ipv6 link local address with a specified scope id
   fe80::aaaa:bbbb:cccc:dddd%eth0 and connects via Http on port 80.
   The RTSP server receives the connection and returns the address
   in the x-server header. The client will receive this address and
   try to connect to it "as is" without the scope id and fail.

In the case of streaming data from RTSP servers like 1. and 2. it's
useful to have the option to simply ignore the x-server header reply
and continue using the original address.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1007>
2021-08-17 10:15:27 +00:00
Sebastian Dröge
a14f4f48c4 video-overlay-composition: Allow empty overlay compositions
Allowing to pass NULL to the constructor removes the need to
special-case the first rectangle in calling code and generally
simplifies application code.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1256>
2021-08-16 21:13:27 +00:00
Matthew Waters
18314764fc webrtc: improve matching on the correct jitterbuffer
The mapping between an RTP session and the SDP m= line is not always the
same, especially when BUNDLEing is used.

This causes a failure in a specific case where if when bundling,
if mline 0 is a data channel, and mline 1 an audio/video section,
then retrieving the transceiver at mline 0 (rtp session used) will fail
and cause an assertion.

This fix is actually potentially a regression for cases where the remote
part does not provide the a=ssrc: media level SDP attributes as is now
becoming common, especially when simulcast is involved.

The correct fix actually requires reading out header extensions as used
with bundle for signalling in the actual data, what media and therefore
transceiver is being used.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2467>
2021-08-16 16:15:44 +00:00
Dmitry Shusharin
a92c855dd5 gstqmlgl: fix indent
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1032>
2021-08-16 11:25:58 +00:00
Dmitry Shusharin
a338ed98d6 gstqmlgl: wrap raw GstGLContext into GWeakRef
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1032>
2021-08-16 11:25:58 +00:00
Dmitry Shusharin
b8cb9ae526 gstqmlgl: add multisink test application
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1032>
2021-08-16 11:25:58 +00:00
Dmitry Shusharin
0bb37c5135 gstqmlgl: refactoring: rename ambiguous variables, clean up unused and duplicated ones
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1032>
2021-08-16 11:25:58 +00:00
Dmitry Shusharin
5dca098f6a gstqmlgl: rework WGL-specific context init code
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1032>
2021-08-16 11:25:58 +00:00
Dmitry Shusharin
83dbeac150 gstqmlgl: retrieve correct device bound to current GL context (+ minor code cleanup)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1032>
2021-08-16 11:25:58 +00:00
Dmitry Shusharin
38b26c2f3f gstqmlgl: correct validation for Qt GL context
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1032>
2021-08-16 11:25:58 +00:00
Dmitry Shusharin
211aaaf8b8 gstqmlgl: create helper QRunnable-based class for render jobs
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1032>
2021-08-16 11:25:58 +00:00
Tulio Beloqui
9af6ce974a rtpjitterbuffer: fixed stall on gap when using rtx
Co-authored-by: Håvard Graff <havard@pexip.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1055>
2021-08-16 09:51:05 +00:00
Per Förlin
535c02c73b gstrtspconnection: Add support to ignore x-server header reply
When connecting to an RTSP server in tunnled mode (HTTP) the server
usually replies with a x-server header. This contains the address
of the intended streaming server. However some servers return an
"invalid" address. Here follows two examples when it might happen.

1. A server use Apache combined with a separate RTSP process to handle
   Https request on port 443. In this case Apache handle TLS and
   connects to the local RTSP server, which results in a local
   address 127.0.0.1 or ::1 in the x-server reply. This address is
   returned to the actual RTSP client in the x-server header.
   The client will receive this address and try to  connect to it
   and fail.

2. The client use a ipv6 link local address with a specified scope id
   fe80::aaaa:bbbb:cccc:dddd%eth0 and connects via Http on port 80.
   The RTSP server receives the connection and returns the address
   in the x-server header. The client will receive this address and
   try to connect to it "as is" without the scope id and fail.

In the case of streaming data from RTSP servers like 1. and 2. it's
useful to have the option to simply ignore the x-server header reply
and continue using the original address.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1192>
2021-08-16 09:06:37 +00:00
He Junyan
fbf6bfd4d8 va: Use GST_CAPS_FEATURE_MEMORY_VA to replace "memory:VAMemory".
"memory:VAMemory" is a commonly used string which notates our VA-kind
memory type. We now used a definition in va lib to replace the simply
string usage.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2466>
2021-08-16 16:25:15 +08:00
He Junyan
d14e8055ad va: Use MEMORY_DMABUF definition to replace "memory:DMABuf" strings.
GST_CAPS_FEATURE_MEMORY_DMABUF is already a common definition, we should
just use it rather than use the "memory:DMABuf" strings by ourselves.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2466>
2021-08-16 16:24:14 +08:00
Nirbheek Chauhan
620e9323c5 flv: use g_memdup2() as g_memdup() is deprecated
g_memdup() is deprecated since GLib 2.68 and we want to avoid
deprecation warnings with recent versions of GLib.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1052>
2021-08-16 08:00:53 +00:00
Sebastian Dröge
04963cac86 souphttpsrc: Always use the content decoder but set Accept-Encoding: identity if no compression should be used
Some servers respond with gzip-encoded responses regardless of whether
the request allowed it to be used in the response. By always having the
content decoder enabled, these invalid responses can be decoded
correctly while for well-behaving servers the `compress` property
selects between allowing compressed responses or not.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/833

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1053>
2021-08-15 12:29:06 +03:00
Mathieu Duponchelle
ee35cba6e8 launcher: don't start the pipeline before we're done updating it
Since 70e3b8ae2a the CommandLineFormatter
also emit "loaded" so we ended up doing this twice, once
as before in `run_pipeline` and another time in the `project:loaded`
callback.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-editing-services/-/merge_requests/265>
2021-08-14 18:07:56 +02:00
Mathieu Duponchelle
e12b3b7cef ges-launcher: don't unref transfer none objects
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-editing-services/-/merge_requests/265>
2021-08-14 18:07:56 +02:00
Piotrek Brzeziński
a5a793f8b6 clip: Copy trackelement's metadata upon splitting
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-editing-services/-/merge_requests/260>
2021-08-14 15:46:07 +02:00
Piotrek Brzeziński
951e6181ce xml-formatter: Add support for metadata on sources
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-editing-services/-/merge_requests/260>
2021-08-14 00:10:06 +02:00
Piotrek Brzeziński
9c03f99e58 marker-list: Add flags (de)serialization
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-editing-services/-/merge_requests/260>
2021-08-14 00:10:06 +02:00
Thibault Saunier
a917648be3 fdkaacdec: Add Converter class to hint gst-validate
fdkaacdec have minimal conversion capability, adding the Converter class allow
gst-validate to behave properly and not spit an error when it notice that the
number of channels or rate miss-match in and out.

Same logic as with opusdec, see: https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1142>

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2462>
2021-08-13 15:25:16 +00:00
Nirbheek Chauhan
e9d06cec5a gstbuffer: Use g_memdup2 instead of g_memdup
This was added in !826 which was created after !803 (which changes
g_memdup -> g_memdup2), but merged before it, so it slipped through.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/866>
2021-08-13 19:55:42 +05:30
Nirbheek Chauhan
3ced923da5 sdp: Avoid using g_memdup() since it is deprecated
g_memdup() is deprecated since GLib 2.68 and we want to avoid
deprecation warnings with recent versions of GLib. Instead of using
g_memdup2(), we can simply use the new gst_buffer_new_memdup() added
in 1.19.x

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1254>
2021-08-13 19:35:23 +05:30
Seungha Yang
b1dd20d57a wasapi2: Increase rank to primary + 1
wasapi2 plugin should be preferred than old wasapi plugin if available because:
* wasapi2 supports automatic stream routing, and it's highly recommended
  feature for application by MS. See also
  https://docs.microsoft.com/en-us/windows/win32/coreaudio/automatic-stream-routing
* This implementation must be various COM threading issue free by design
  since wasapi2 plugin spawns a new dedicated COM thread and all COM objects'
  life-cycles are managed correctly.
  There are unsolved COM issues around old wasapi plugin. Such issues are
  very tricky to be solved unless old wasapi plugin's threading model
  is re-designed.

Note that, in case of UWP, wasapi2 plugin's rank is primary + 1 already

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2314>
2021-08-13 12:35:11 +00:00
Mathieu Duponchelle
152813e71d ccconverter: fix overflow when not doing framerate conversion
When converting from one framerate to another, counters are
reset periodically, however when not converting they never are
and can_genearte_output ends up making overflow-prone calculations
with large values for input_frames and output_frames.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2465>
2021-08-13 03:37:28 +00:00
Xavier Claessens
ebcca1e5ea Meson: Avoid using add_global_arguments() when gst-build is a subproject
Meson only allows the main project to use it. We already set that flag
in all GStreamer modules just like warning flags.

Fixes: #152
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/236>
2021-08-12 14:17:30 +00:00
Thibault Saunier
c24958c781 git-update: Fix passing fetch_args
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/255>
2021-08-12 13:20:15 +00:00
Thibault Saunier
b29665f3bd git-update: Force fetching tags
Making it simpler for user to specify tags in manifests

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/255>
2021-08-12 13:20:15 +00:00
Matthew Waters
cfd4a9a6d9 qt: always update the sink_retrieved flag when the sink retrieves
Fixes a case where adding a qmlgloverlay element after an existing
qmlglsink elements was already in the pipeline would create an entirely
separate GstGLDisplay pointing to the same underlying display resource.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1050>
2021-08-12 22:57:01 +10:00
Sebastian Dröge
01c430fa45 webrtcbin: Don't assume that non-audio medias are video medias when creating transceivers
And print the unknown media kind in the logs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2464>
2021-08-12 12:31:15 +00:00
Sebastian Dröge
7a03acc546 webrtcbin: Use the correct media for deciding the media kind when creating the transceiver from the SDP
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2464>
2021-08-12 12:31:15 +00:00
He Junyan
70ce2327d0 codecs: h264dec: Output the picture directly if already a frame.
We forget one case that is the frame and field pictures may be mixed
together. For this case, the dpb is interlaced while the last picture
may be a complete frame. We do not need to cache that complete picture
and should output it directly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2448>
2021-08-12 07:41:28 +00:00
Víctor Manuel Jáquez Leal
862aa25e53 videocrop: Resurrect logging category.
Fix for a regression from commit 8f1384c9. That commit moved the debug
category definition, as static, into a gstvideocropelement.c, but that
category was used as default, in gstvideocrop.c, so it was never used
at logging, so the debug selector never showed the logs for
videocrop.

This patch move back the category definition into gstvideocrop.c and
leaving the function videocrop_element_init() as a noop.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1049>
2021-08-11 16:09:06 +02:00
He Junyan
2505ab17e9 va: caps: Make the template raw video caps classified by features.
The current output of raw video caps is not good. When we have multi
profiles and each profile support different formats, the output of
gst-inspect may like:

 SRC template: 'src'
 Availability: Always
 Capabilities:
   video/x-raw(memory:VAMemory)
         width: [ 1, 16384 ]
         height: [ 1, 16384 ]
         format: NV12
   video/x-raw
         width: [ 1, 16384 ]
         height: [ 1, 16384 ]
         format: NV12
   video/x-raw(memory:VAMemory)
         width: [ 1, 16384 ]
         height: [ 1, 16384 ]
         format: P010_10LE
   video/x-raw
         width: [ 1, 16384 ]
         height: [ 1, 16384 ]
         format: P010_10LE
   video/x-raw(memory:VAMemory)
         width: [ 1, 16384 ]
         height: [ 1, 16384 ]
         format: P012_LE
   video/x-raw
         width: [ 1, 16384 ]
         height: [ 1, 16384 ]
         format: P012_LE

The gst_caps_simplify does not classify the caps by same features, but
just leave them interweaved. We need to handle them manually here, the
result should be:

  SRC template: 'src'
  Availability: Always
  Capabilities:
    video/x-raw
          width: [ 1, 16384 ]
          height: [ 1, 16384 ]
          format: { (string)P010_10LE, (string)P012_LE, (string)NV12 }
    video/x-raw(memory:VAMemory)
          width: [ 1, 16384 ]
          height: [ 1, 16384 ]
          format: { (string)P010_10LE, (string)P012_LE, (string)NV12 }

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2456>
2021-08-11 09:37:33 +00:00
Víctor Manuel Jáquez Leal
f20b3b8156 vapostproc: Inherit from GstVaBaseTransform.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2442>
2021-08-10 17:31:58 +00:00
Víctor Manuel Jáquez Leal
977a8f3b01 va: Add base transform class.
This base transform class is a derivable class for VA-based filters,
for example vapostproc right now, but it will be used also for
future elements such as vadeinterlace.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2442>
2021-08-10 17:31:58 +00:00
Víctor Manuel Jáquez Leal
2added54c3 va: pool: Add gst_va_pool_new_with_config().
It is a function helper.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2442>
2021-08-10 17:31:58 +00:00
Seungha Yang
1f6fd7550c d3d11window: Misc code cleanup
* Remove unnecessary upcasting. We are now dealing with C++ class objects
  and don't need explicit C-style casting in C++ world
* Use helper macro IID_PPV_ARGS() everywhere. It will make code
  a little short.
* Use ComPtr smart pointer instead of calling manual IUnknown::Release()

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2461>
2021-08-10 16:20:37 +00:00
Seungha Yang
a1048ce110 d3d11compositor: Fix indent
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2461>
2021-08-10 16:20:37 +00:00