Commit graph

18855 commits

Author SHA1 Message Date
Sebastian Dröge
be1c97d3c9 qtdemux: Add a minimum buffer size for raw audio to not output one buffer per frame
Outputting 48000 buffers per second is not a good idea performance-wise.
If a container sample is less than 1024 raw audio frames, combine
multiple samples to get at least 1024 raw audio samples as long as
they're stored contiguous in the file.

For the other direction, if a container sample contains more than 4096
samples there is already code for splitting them up.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692750
2020-02-14 08:48:01 +00:00
Mathieu Duponchelle
1471100f37 rtspsrc: fix requested range
When the server replies with a range "now-", it is presumed to
be a "live" stream and we should request a similar range.

This was the case prior to my refactoring to make use of
gst_rtsp_range_to_string in 5f1a732bc7,
this commit restores the behaviour for that case.
2020-02-12 05:47:54 +00:00
Mikhail Fludkov
57b0522cd7 rtpptdemux: set payload to caps inside gst_rtp_pt_demux_get_caps
Refactoring to remove duplicate code and add test
2020-02-11 18:39:22 +00:00
Stian Selnes
629b71ac9c rtpptdemux: Fix debug to use GST_DEBUG_OBJECT 2020-02-11 18:39:22 +00:00
Mikhail Fludkov
851a2b7925 rtpbin: use max-streams on rtpssrcdemux
The proper way of capping on max-streams is to do it in rtpssrcdemux.
This patch uses the newly introduced property on rtpssrcdemux. Previous
behavior would not prevent rtpssrcdemux spawning new pads for every new
ssrc and potentialy causing performance trouble during teardown.
2020-02-11 15:12:07 +01:00
John Bassett
16d750bc01 rtpssrcdemux: Handle RTCP APP packets
Fix crash when processing RTCP APP packets.
2020-02-11 15:12:07 +01:00
John Bassett
5800950a2d rtpssrcdemux: Bad RTP/RTCP packet is not fatal
When used for processing bundled media streams within rtpbin the rtpssrcdemux element may
receive bad RTP and RTCP packets, these should not be treated as a fatal error.
2020-02-11 15:10:12 +01:00
Mikhail Fludkov
35596e7fac rtpssrcdemux: introduce max-streams property
The property is useful against atacks when the sender changes SSRC for
every RTP packet. The property with the same name introduced in rtpbin
was not enough, because we still can end up with thousands of pads
allocated in rtpssrcdemux.
2020-02-11 15:10:12 +01:00
Havard Graff
94e10d522e rtpssrcdemux: fix test warnings 2020-02-11 15:07:45 +01:00
Alexander Lapajne
54c4ba82f8 rtspsrc: Fix for segmentation fault when handling set/get_parameter requests
gstrtspsrc uses a queue, set_get_param_q, to store set param and get
param requests. The requests are put on the queue by calling
get_parameters() and set_parameter(). A thread which executs in
gst_rtspsrc_thread() then pops requests from the queue and processes
them. The crash occured because the queue became empty and a NULL
request object was then used. The reason that the queue became empty
is that it was popped even when the thread was NOT processing a get
parameter or set parameter command. The fix is to make sure that the
queue is ONLY popped when the command being processed is a set
parameter or get parameter command.
2020-02-10 09:43:17 +01:00
Olivier Crête
c00796eaa5 rtpsession: Add test for packet rate maths 2020-02-06 14:01:38 -05:00
olivier.crete@collabora.com
774ddb15b8 rtpstats: Base the packet rate average on the packet rate itself
Do this so that the average update speed is in time instead of varying
based on the actual packet arrival rate.
2020-02-06 14:00:48 -05:00
olivier.crete@collabora.com
a637ec3da8 rtpstats: Don't save the ts & seqnum if the avg is not updated
This makes it update correctly when you have more than one packet per
frame.
2020-02-06 14:00:48 -05:00
Guillaume Desmottes
48a7381602 v4l2: map GST_VIDEO_FORMAT_BGR15
The GstVideoFormat to v4l2 conversion was missing for BGR15.
2020-02-05 18:22:20 +05:30
Guillaume Desmottes
0f907205de v4l2: fix crash on invalid caps
gst_v4l2_object_set_format_full() was returning FALSE without setting
an error. Caller code (gst_v4l2src_fixate()) was then derefing a
NULL pointer when trying to handle the error.
2020-02-05 18:22:20 +05:30
Sebastian Dröge
f6e383b749 splitmuxsink: Include actual sink element in the fragment-opened/closed messages
If not configuring the sinks via the "location" property this can be
useful to know for which sink the fragment was actually opened/closed,
especially if finalization of the fragments is happening asynchronously.
2020-01-29 13:30:00 +00:00
Juergen Werner
755dba4561 rtpjitterbuffer: fix scaling from RTP-time to NTP-time
The scaling was inverse.
2020-01-29 12:05:07 +01:00
Mathieu Duponchelle
a245e85fb1 rtprtxsend: allow generic input caps
When connected to an upstream rtpfunnel element, payload-type,
ssrc and clock-rate will not be present in the received caps.

rtprtxsend can already deal with only the clock rate being
present there, a new property is exposed to allow users to
provide a payload-type -> clock-rate map, this enables the
use of the max-size-time property for bundled streams.
2020-01-28 15:44:13 +00:00
Julien Isorce
88ce7397fc vp8enc/vp8enc: set 1 for the default value of VP8E_SET_STATIC_THRESHOLD
In Google webrtc, the setting VP8E_SET_STATIC_THRESHOLD is set to 1
(except when the content is known to be static very often in which
case it is set to 100, i.e. when sharing screen with Google Hangouts).

The cpu usage drops a lot when using 1 for above setting because it
allows the encoder to skip static/low content blocks. The current
0 default value uses too much cpu and confuses the user regarding
the cpu usage expectations. User expects vp8enc to use low cpu by
default.

Documentation of VP8E_SET_STATIC_THRESHOLD:
  https://github.com/webmproject/libvpx/blob/master/vpx/vp8cx.h#L188

chromium/webrtc:
  b484ec0082/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc (822)

Closes #58
2020-01-28 02:41:50 +00:00
Nicolas Dufresne
83e9d4f70d jpegdec: Check return value of gst_buffer_map()
Without this check, the element will crash instead of returning an
error.
2020-01-27 22:59:34 +00:00
Sebastian Dröge
eb0b676fae splitmuxsink: Check the correct sink class for the existence of the "location" property 2020-01-27 15:53:40 +02:00
Sebastian Dröge
5877d945a4 qtdemux: Always prefer information from v1/v2 sound sample description over sample description entry
ffmpeg is doing the same and various files in the wild have bogus
information in the sample description if the same information is also
duplicated afterwards in the v1/v2 sound sample desription.

Previously we only did this for non-raw audio due to
  https://bugzilla.gnome.org/show_bug.cgi?id=374914
but this specific file is already worked around differently. It still
works after this change.

Also remove ad-hoc GST_READ_DOUBLE_BE re-implementation and move the
switch for legacy audio formats after reading all the sample
descriptions as we want to override the values from there.
2020-01-27 14:14:50 +02:00
Sebastian Dröge
c4f6ce789d avimux: Add support for >2 raw audio channels
For this case write a WAVEFORMATEXTENSIBLE header and also reorder the
raw audio channels to the AVI channel order if needed.
2020-01-19 12:09:38 +00:00
Sebastian Dröge
451fc5c112 wavenc: Fix writing of the channel mask with >2 channels
The channel position is an enum but the conversion code assumed it's a
mask. Convert accordingly.
2020-01-13 19:50:06 +00:00
Kristofer Björkström
9c86414279 rtph265pay: TID for NALU type 48 was always set to 7
A typo bug: | instead of & resulted in TID alwasy being set to 7
for the aggregated NALU of type 48
2020-01-13 15:41:30 +01:00
Sebastian Dröge
c17d5e36ad imagefreeze: Add support for replacing the output buffer
By default imagefreeze will still reject new buffers after the first one
and immediately return GST_FLOW_EOS but the new allow-replace property
allows to change this.

Whenever updating the buffer we now also keep track of the configured
caps of the buffer and from the source pad task negotiate correctly
based on the potentially updated caps.

Only the very first time negotiation of a framerate with downstream is
performed, afterwards only the caps themselves apart from the framerate
are updated.
2020-01-11 08:04:43 +00:00
Alicia Boya García
8dd42666e3 qtdemux: Fix race on pad reconnection
Elements emitting frames through several srcpads should use a
flow combiner to aggregate the chain returns and therefore only return
GST_FLOW_NOT_LINKED to upstream when all the downstream pads have
received GST_FLOW_NOT_LINKED.

In addition to that, in order to handle pads being relinked downstream,
the flow combiner should be reset in response to RECONFIGURE events.
This ensures that a both srcpads process a chain operation before a
GST_FLOW_NOT_LINKED can be propagated upstream (which would usually stop
the pipeline).

Otherwise, in a configuration with two srcpads, only one linked at a
time, after the relink the element could chain data through the now
unlinked pad and the flow combiner would resolve as GST_FLOW_NOT_LINKED
(stopping the pipeline) just because the now linked pad has not been
chained yet to update the flow combiner.

This patch adds handling of RECONFIGURE events to qtdemux. Also, since
this event handling causes the flow combiner to be used from a thread
other than the qtdemux streaming thread, usages of the flow combiner
has been guarded by the object lock.
2020-01-09 18:43:02 +00:00
Seungha Yang
8445685a21 splitmuxsink: Fix assertion failure on set_property()
GValue might have null object.

(gst-inspect-1.0:10304): GStreamer-CRITICAL ...
    gst_object_ref_sink: assertion 'object != NULL' failed
2020-01-07 01:24:01 +09:00
Daniel Molkentin
bb1ce82e39 videocrop: allow properties to be animated by GstController 2020-01-03 15:16:02 +01:00
Aaron Boxer
09d4514814 rtspsrc: improved handling of control concatenation with base
Also, `control_url` variable has been renamed to `control_path`,
as it is actually a path.
2019-12-30 16:52:45 +00:00
Aaron Boxer
ed6b5a3a63 rtspsrc: append aggregate control string to base URL before query string
Appending control string to end of query changes meaning of query string
Fixes #650
2019-12-30 16:52:45 +00:00
Eric Marks
d6961235e8 aasink & cacasink: add filter aatv & cacatv
Add transform filter capabilities to aasink and cacasink in the form of new elements aatv and cacatv.
2019-12-28 23:01:19 +00:00
Niels De Graef
acab06b2e8 alpha: Cleanup using G_DECLARE_FINAL_TYPE
We started depending on GLib 2.44, so we can clean up all the GObject
boilerplate macros.
2019-12-28 04:05:13 +00:00
Stéphane Cerveau
b928517f1e good: use of g_value_dup_string
Use helper method to get string from GValue.
2019-12-20 09:30:26 +00:00
Havard Graff
8b96d8ee8d rtpbin: fix shutdown crash in rtpbin
The key is to make sure the jitterbuffer is set to NULL *before* the
ptdemux.

The race that existed would basically happen when ptdemux had reached
READY, and the jitterbuffer would then push a buffer, triggering a new
pad with a new payloadtype being added and ghosted to the rtpbin itself.

However, the srcpad of the ptdemux would now be inactive, and all the
sticky-event pushed on it would be swallowed, not allowing any to reach
the ghost-pad. Then the buffer in-flight would come to the ghostpad,
and we would assert that a buffer arrived before the necessary
events.

By simply re-ordering the state-changes, we ensure that there will be
no buffer racing into the ptdemux while its state is being changed,
and the problem disappears completely.

Notice also that there is not point in disconnecting the signals on the
ptdemux before this point, since we need the push-thread to settle
down before we can do this in a non-racy way.
2019-12-20 08:27:07 +00:00
Aaron Boxer
4155c59cc4 rtspsrc: avoid seek DISCONT when only rate changes in same direction
Not setting DISCONT avoids a noticable delay when seeking
with only rate changing, in the same direction as current
rate.
2019-12-19 05:54:38 +00:00
Olivier Crête
9db1d740e8 rtspsrc: Remove deprecated GTimeVal
GTimeVal won't work past 2038
2019-12-18 19:48:34 +00:00
Olivier Crête
f66fc2a694 osxaudio: Remove deprecated GTimeVal 2019-12-18 19:48:34 +00:00
Sebastian Dröge
04806a75bd avimux: Add support for S24LE and S32LE raw audio
avidemux already handles this correctly.
2019-12-18 11:16:30 +00:00
Sebastian Dröge
4dbaff424f avimux: Allow muxing v210 video into AVI
avidemux already handles this.
2019-12-18 10:20:25 +00:00
Vivia Nikolaidou
7cbc351e05 flvdemux: Don't replace video codec data when we receive a PAR
Receiving a pixel-aspect-ratio should trigger a caps change, but not
replace the existing video codec tag
2019-12-16 21:51:38 +00:00
Mathieu Duponchelle
5766731bd4 qtmux: protect access to GstElement.sinkpads 2019-12-16 14:17:38 +00:00
Mathieu Duponchelle
e2462005fb qtmux: port to GstAggregator 2019-12-16 14:17:38 +00:00
Joakim Johansson
4d7d577496 gstrtspsrc: Add missing lock on free set_get_param_q
Otherwise is it possible to get a crash in gst_rtspsrc_set_parameter.
2019-12-16 13:13:00 +01:00
Sebastian Dröge
9f6ed9ec72 splitmuxsink: Increment fragment_id even if no fragment location was provided
Applications might handle locations and generally configuration of the
sink by themselves instead of having splitmuxsink set the location on
the sink. Nonetheless it makes sense to increment the fragment_id that
is passed to the signal so that applications know which fragment is
requested.
2019-12-13 22:59:55 +00:00
Jan Alexander Steffens (heftig)
9e0eb77810
flvmux: Use the last DTS for the metadata timestamp
This avoids creating a timestamp regression during a stream.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/merge_requests/429
2019-12-12 11:09:31 +01:00
Mathieu Duponchelle
625eb00c06 qtdemux: send GAP events for lagging audio and video streams too
The logic is taken straight from matroskademux, see
77403d0afe
2019-12-11 19:59:13 +00:00
Seungha Yang
5009cad220 flvmux: Use thread-safe gmtime_r if available
gmtime on *nix is not thread-safe.
2019-12-10 23:48:35 +09:00
Stéphane Cerveau
b44d37a338 splitmuxsink: provides a start-index property
Allow to change the fragment-id start index.
2019-12-05 14:58:40 +00:00
Philipp Zabel
c47a585052 qmlglsink: fix build on EGL platform without X11 headers
If Mesa is built without X11 headers, building against Mesa EGL headers
requires a dependency on egl.pc, to define MESA_EGL_NO_X11_HEADERS.
This fixes a build error when compiling ext/qt/gstqtglutility.cc:

  In file included from /usr/include/EGL/egl.h:39,
                   from /usr/include/gstreamer-1.0/gst/gl/egl/gstegl.h:44,
                   from ../gst-plugins-good-1.16.1/ext/qt/gstqtglutility.cc:43:
  /usr/include/EGL/eglplatform.h:124:10: fatal error: X11/Xlib.h: No such file or directory
2019-12-04 22:38:49 +00:00