Commit graph

18572 commits

Author SHA1 Message Date
Jochen Henneberg
88a6832b2a flacdec: Use new channel count for audio info 2018-12-14 09:22:33 +00:00
Jochen Henneberg
d74f964407 flacdec: Caps may have changed on FLAC metadata change
If the decoder signals metadata change we need to update the output
format and negotiate with downstream elements.
2018-12-14 09:22:33 +00:00
Jochen Henneberg
a5d71c597d flacdec: Reset decoder on set_format()
Any call to set_format() could mean that the stream type changed so we
reset the decoder and mark got_headers FALSE.
2018-12-14 09:22:33 +00:00
Jochen Henneberg
7824e87c5b flacparse: On sink caps change restart parser
Draining the parser is not enough here, on caps change we need to
reset it so it is ready to accept new caps.
2018-12-14 09:22:33 +00:00
Jochen Henneberg
9b6dcc7f1b rtpgstdepay: Update pad caps if inline caps change
If the inlined caps change while using the same CV we need to update the
source pad caps.
2018-12-14 09:22:33 +00:00
Justin Kim
34233baae1 osxvideo: meson: Add dependencies by using appleframeworks
Otherwise, it fails to link.

gst-build#13
2018-12-14 12:21:58 +09:00
Sebastian Dröge
172d14acef cairooverlay: Optimize premultiplication/unpremultiplication loops
Pull in video frame fields into local variables. Without this the
compiler must assume that they could've changed on every use and read
them from memory again.

This reduces the inner loop from 6 memory reads per pixels to 4, and the
number of writes stays at 3.
2018-12-07 17:22:39 +00:00
Sebastian Dröge
c50be8f146 qtdemux: Put framerate into the closedcaption caps if it can be calculated from the stream
Using the same calculation used for video streams.
2018-12-06 16:05:50 +00:00
Sebastian Dröge
830e7dc14b qtmux: Set timescale of closedcaption tracks to the one of the main video track 2018-12-06 16:05:50 +00:00
Thibault Saunier
0adac97e1c Automatic update of common submodule
From ed78bee to 59cb678
2018-12-05 17:24:13 -03:00
Maciej Wolny
ec655de288 Remove duplicate declarations
This causes 'redefinition of typedef ...' errors on GCC 4.5.3
2018-12-04 11:13:02 +00:00
Tim-Philipp Müller
193dbd20a8 tests: rtpssrcdemux: fix uninstalled autotools build and distcheck 2018-11-30 23:56:12 +00:00
Alicia Boya García
38b553dda7 qtdemux: set need_segment after a second moov
stream.segment should be updated with the values of the current edit
list, also when a new `moov` is received. Unfortunately this was not
being the case because of an early return.

As a consequence of this bugs, no end of movie clipping was being
performed on the new moov and no segment event was being emitted.

When performing stream switching (e.g. in MSE) the new moov may have a
different edit list. This is often the case when switching between
baseline H.264 (which lacks B-frames) and more demanding profiles. For
this reason it's important to emit a new segment in order to be able
to get matching stream times.
2018-11-30 20:44:57 +00:00
Alicia Boya García
26cc201c8a qtdemux: Initialize QtDemuxStream.segment in its constructor
This patch moves the initialization of QtDemuxStream.segment from
gst_qtdemux_add_stream() to _create_stream(). This ensures the segment
is always initialized when the stream is created.

Otherwise the segment format is left as GST_FORMAT_UNDEFINED in the case
were a track is reparsed and qtdemux_reuse_and_configure_stream() is
called instead of gst_qtdemux_add_stream(). (See
qtdemux_expose_streams() in the non streams-aware case.)
2018-11-30 20:44:57 +00:00
Miguel Paris
48a4fd4e50 rtpsession: properly handle rtcp_feedback_retention_window
- Consider GST_CLOCK_TIME_NONE as not to be used.
- Complete "rtcp-feedback-retention-window" property getter/setter
  implementation.
2018-11-30 10:55:26 +00:00
Miguel Paris
458741e4b2 rtpsource: properly prune RTCP packets out of feedback_retention_window
Closes #522
2018-11-30 10:55:26 +00:00
Miguel Paris
53f03d4cc1 rtpsource: properly compare buffer PTSs 2018-11-30 10:55:26 +00:00
Miguel Paris
57829c3352 rtpsource: retain_rtcp_packet: warning if invalid running_time 2018-11-30 10:55:26 +00:00
Miguel Paris
36f55b03e8 rtpsession: properly set the running_time for rtcp packet info 2018-11-30 10:55:26 +00:00
Nicolas Dufresne
d637567ab3 rtpssrcdemux: Rename confusingly name lock macros
This is an extra internal recurisve lock use to avoid having to take
both sink pad streams lock all the time. This patch renamed it
INTERLNAL_STREAM_LOCK/UNLOCK() to avoid confusion with possible upstream
GST_PAD API.
2018-11-29 15:34:47 -05:00
Nicolas Dufresne
40daf6322d rtpssrcdemux: Hold on internal stream lock while pushing sticky
This reverts "6f3734c305 rtpssrcdemux: Only forward stick events while
holding the sinkpad stream lock" and actually hold on the internal
stream lock. This prevents in some needed case having a second
streaming thread poping in and messing up event ordering.
2018-11-29 15:33:57 -05:00
Nicolas Dufresne
c596bdda38 test: rtpssrcdemux: Test event forwarding
This the first unit test of this element. It adds a test that verify
that events are forwarded correctly.
2018-11-29 15:19:17 -05:00
Matej Knopp
e9495c55f4 matroskademux: fix handling of MS ACM audio
Pass riff codec-data as strf, not strd, which is where
gst_riff_create_audio_caps() expects the WAVEFORMATEXTENSIBLE
data.

https://bugzilla.gnome.org/show_bug.cgi?id=757583
Fixes #234
2018-11-28 11:55:14 +00:00
Jordan Petridis
515ada7e22
Run gst-indent through the files
This is required before we enabled an indent test in the CI.

https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/33
2018-11-28 05:52:16 +02:00
Thibault Saunier
2e0a45d7df aspectcropration: Fix potential unref of NULL pointer 2018-11-26 08:11:57 -03:00
Thibault Saunier
eb2b58cc0b aspectcropratio: Set caps from the streaming thread on property changes
Otherwise it might lead to deadlocks

See https://gitlab.gnome.org/GNOME/pitivi/issues/2259

Closes #518
2018-11-26 07:14:09 -03:00
Nicolas Dufresne
21378d83c2 rtpssrcdemux: Forward serialized events to all pads
While forwarding serialized event, we use gst_pad_forward() function.
In the forward callback (GstPadForwardFunction) we always return
TRUE. Returning true there will stop the dispatching procedure. As a
side effect, only one events is receiving the events. This breaks
when sending EOS from the applicaiton, it also breaks the latency
tracer.
2018-11-24 13:01:25 +00:00
Seungha Yang
69ad6af83c meson: Specify encoding to UTF-8 when building with MSVC
Use build arguments consistent with core and -base. This can also
remove noisy "C4819" warning of non-us locale MSVC.
2018-11-24 20:16:30 +09:00
Xavier Claessens
5fad0735a5 Check for zlib header 2018-11-22 20:04:53 -05:00
Nicolas Dufresne
3b22a604b5 v4l2: Properly fix Android build
The previous patch did not even compile on any possible platform or C
standard. That commit also didn't have a proper commit message.

Android ships Linux with a different signature for ioctl. They first
released an ioctl with int as request type, and later "fixed" it by
adding an override with unsign, which is still not matching Linux and
BSD implementation which uses unsigned long int.
2018-11-21 18:58:38 -05:00
Xavier Claessens
0e2d0adac5 Fix ioctl() signature on Android 2018-11-21 16:11:02 -05:00
Xavier Claessens
b93e37592a Fix zlib detection when there is no pkg-config file 2018-11-19 15:21:07 -05:00
Arun Raghavan
a24ef929a4 pulse: Expose the correct max rate that we support
PulseAudio defines PA_RATE_MAX as the maximum sampling rate that it
supports. We were previously exposing a maximum rate of INT_MAX, which
is incorrect, but worked because nothing was really using a rate greater
than 384000 kHz.

While playing DSD data, we hit a case where there might be very high
sample rates (>1MHz), and pulsesink fails during stream creation with
such streams because it erroneously advertises that it supports such
rates.

Since PA_RATE_MAX is #define'd to (8*48000U), we can't just use it in
the caps string. Instead, we fix up the rate to what we actually support
whenever we use our macro caps.
2018-11-19 21:28:25 +05:30
Alicia Boya García
753b7c17f3 matroskademux: Defer seeks received before GST_MATROSKA_READ_STATE_DATA
This patch enables matroskademux to receive seeks before it reaches
GST_MATROSKA_READ_STATE_DATA.

Closes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/514

This also enables receiving seeks in the element READY state.

When such a seek is received, it is stored to be later handled when
GST_MATROSKA_READ_STATE_DATA is reached.
2018-11-15 08:01:29 +00:00
Linus Svensson
8fc8b7ee33 rtpsession: Implement reset
Reset RTPSession when rtpsession changes state from PAUSED to READY.
Without this change, a stored last_rtptime in RTPSource could interfere
with RTP timestamp generation in RTCP Sender Report.

Fixes #510
2018-11-13 12:30:35 +00:00
Linus Svensson
ac94c706da rtpsession: test: Plug memory leak 2018-11-13 12:30:35 +00:00
Mathieu Duponchelle
fd560bcb27 rtpfunnel: Stop using G_DECLARE_FINAL_TYPE
Fixes #516
2018-11-13 00:37:11 +01:00
Jordan Petridis
c9ed35f641 Add Gitlab CI configuration
This commit adds a .gitlab-ci.yml file, which uses a feature
to fetch the config from a centralized repository. The intent is
to have all the gstreamer modules use the same configuration.

The configuration is currently hosted at the gst-ci repository
under the gitlab/ci_template.yml path.

Part of https://gitlab.freedesktop.org/gstreamer/gstreamer-project/issues/29
2018-11-12 11:43:20 +00:00
Nicolas Dufresne
dbaca549ab v4l2object: Read driver selected interlace mode
If there was no interlace-mode field in the caps. Read back the value
selected by the driver. This way, if the driver does not support
progressive, then it will automatically negotiate the returned mode
unless this mode is not supported by GStreamer.

This method was already used for colorimetry. Just like colorimetry, the
interlace mode is not longer probed by v4l2src dues to performance
issues.

Fixes #511
2018-11-07 19:29:51 -05:00
Matthew Waters
40fc8aea8f matroska: implement preliminary support for the bitrate query
Return the size / total duration as a ballpark estimate.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/60
2018-11-07 15:07:18 +00:00
Matthew Waters
8a7074f748 isomp4: add preliminary support for the bitrate query
Return the upstream size over the duration as a first estimate.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/60
2018-11-07 15:07:18 +00:00
Sebastian Dröge
87202cc03d rtpbin: Sink jitterbuffer/storage before passing as parameters to signals
Otherwise signal handlers from bindings will take ownership of them as
they are still floating, and we won't own a reference inside rtpbin
anymore.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/515
2018-11-07 09:11:16 +00:00
Havard Graff
65a7d39bd4 flvmux: Test that timestamps are always increasing
Decreasing timestamps break rtmpsink.

With contributions from Olivier Crête.

https://bugzilla.gnome.org/show_bug.cgi?id=796382
2018-11-05 18:17:04 -05:00
Olivier Crête
fea0d0b1a4 flvmux: Force timestamps to always be increasing
https://bugzilla.gnome.org/show_bug.cgi?id=796382
2018-11-05 18:17:01 -05:00
Matthew Waters
dd5c6fe330 Update common submodule location
Remove the git directory
2018-11-05 05:36:26 +00:00
Haihao Xiang
39594d7155 Clone the code from gitlab
This fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/513
2018-11-05 12:26:10 +08:00
Seungha Yang
5d542030db qtdemux: Ignore corrupted CTTS box
If ctts (CompositionOffsetBox) has larger sample_offset
(offset between PTS and DTS) than (2 * duration) of the stream,
assume the ctts box to be corrupted and ignore the box.

https://bugzilla.gnome.org/show_bug.cgi?id=797262
2018-11-01 16:03:12 +02:00
Sebastian Dröge
a03d29420b scaletempo: Implement SEGMENT query
https://bugzilla.gnome.org/show_bug.cgi?id=797313
2018-10-28 17:52:18 +00:00
Sebastian Dröge
2415d517f1 wavparse: Implement SEGMENT query
https://bugzilla.gnome.org/show_bug.cgi?id=797313
2018-10-28 17:52:18 +00:00
Olivier Crête
486044063a dtmfsrc: Declare output as interleaved
This element doesn't support planar audio yet.
2018-10-28 17:12:59 +00:00