Commit graph

18530 commits

Author SHA1 Message Date
Guillaume Desmottes
f0eb1419f6 deinterlace: stop checking line index boundaries
The LINE2() macro already prevents out of bound indexes using CLAMP_HI()
and CLAMP_LOW().
2020-03-03 17:15:00 +00:00
Guillaume Desmottes
cca8008779 deinterlace: fix video info on output frames
Output frames used to have their interlace mode set to the same one as
the input. This breaks their field and comp heights when deinterlacing
an alternate stream.
2020-03-03 17:15:00 +00:00
Guillaume Desmottes
6dde6038cc deinterlace: use output caps to compute buffer size
In interlace-mode=alternate the input buffers have half the size of the
output ones as each field has its own buffer.
2020-03-03 17:15:00 +00:00
Jennifer Berringer
3287f1cb3f flacparse: fix broken reordering of flac metadata
Each FLAC metadata block starts with a flag denoting whether it is the
last metadata block. The existing flacparse code moves any existing
VORBISCOMMENT block to immediately follow the STREAMINFO block without
changing any block's last-metadata-block flag. If no VORBISCOMMENT block
exists, it created one with the last-metadata-block flag set to true.
This results in gstflacdec sometimes giving bad headers to libflac when
trying to play perfectly valid FLAC files depending on the file's
metadata ordering. Depending on the contents of the other metadata
blocks, current versions of libflac may or may not return
FLAC__STREAM_DECODER_ERROR_STATUS_BAD_HEADER when given this broken
metadata. This is most noticeable with files that have a large cover art
image attached where VORBISCOMMENT is the very last metadata block with
no PADDING afterwards.

This patch changes that behavior so that:

1. For FLAC files that already have a VORBISCOMMENT block, the metadata
   order is preserved.
2. For FLAC files that do not have a VORBISCOMMENT block, the generated
   dummy VORBISCOMMENT is placed immediately after STREAMINFO and
   inherits the last-metadata-block flag from STREAMINFO.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/484
2020-03-03 08:03:32 +00:00
Yeongjin Jeong
830db205f6 tests: flvmux: Instead of using the testclock, just send eos event for drain
When using the testclock for determining clock in test, it is sometimes observed
that the clock entry is not registered in time by the aggregator. So deadlock occurs
between the aggregator and the test thread.
2020-03-02 01:37:27 +09:00
Sebastian Dröge
885d330ee6 qtdemux: Try to infer useful header values for raw audio if the sound sample descriptions contain zero values 2020-02-28 13:52:40 +00:00
Sebastian Dröge
9e9af6711d qtdemux: Also use the enda atom for determining endianess of in32, fl32 and fl64 formats
Previously it was only used for in24.
2020-02-28 13:52:40 +00:00
Sebastian Dröge
67be373221 qtdemux: Fix up header information for various fixed-format raw audio formats
Sometimes the headers contain useless, wrong or zero values for e.g. the
sample size with these formats. There's only a single valid value for
them so let's set these instead.
2020-02-28 13:52:40 +00:00
Sebastian Dröge
2c5f6e508c qtdemux: Don't print "unhandled type" warnings for various other raw audio fourccs 2020-02-28 13:52:40 +00:00
Sebastian Dröge
65b30ecce6 qtdemux: Add some more raw audio fourccs to the header instead of duplicating them 2020-02-28 13:52:40 +00:00
Nirbheek Chauhan
42e7864e90 rtpjitterbuffer: Don't use glib format modifiers with sscanf
We do not have a way to know the format modifiers to use with string
functions provided by the system. G_GUINT64_FORMAT and other string
modifiers only work for glib string formatting functions. We cannot
use them for string functions provided by the stdlib. See:
https://developer.gnome.org/glib/stable/glib-Basic-Types.html#glib-Basic-Types.description

```
../gst/rtpmanager/gstrtpjitterbuffer.c: In function 'gst_jitter_buffer_sink_parse_caps':
../gst/rtpmanager/gstrtpjitterbuffer.c:1523:32: error: unknown conversion type character 'l' in format [-Werror=format=]
           || sscanf (mediaclk, "direct=%" G_GUINT64_FORMAT, &clock_offset) != 1)
                                ^~~~~~~~~~
In file included from /home/nirbheek/cerbero/build/dist/windows_x86/include/glib-2.0/glib/gtypes.h:32,
                 from /home/nirbheek/cerbero/build/dist/windows_x86/include/glib-2.0/glib/galloca.h:32,
                 from /home/nirbheek/cerbero/build/dist/windows_x86/include/glib-2.0/glib.h:30,
                 from /home/nirbheek/cerbero/build/dist/windows_x86/include/gstreamer-1.0/gst/gst.h:27,
                 from /home/nirbheek/cerbero/build/dist/windows_x86/include/gstreamer-1.0/gst/rtp/gstrtpbuffer.h:27,
                 from ../gst/rtpmanager/gstrtpjitterbuffer.c:108:
/home/nirbheek/cerbero/build/dist/windows_x86/lib/glib-2.0/include/glibconfig.h:69:28: note: format string is defined here
 #define G_GUINT64_FORMAT "llu"
                            ^
../gst/rtpmanager/gstrtpjitterbuffer.c:1523:32: error: too many arguments for format [-Werror=format-extra-args]
           || sscanf (mediaclk, "direct=%" G_GUINT64_FORMAT, &clock_offset) != 1)
                                ^~~~~~~~~~
```

See also: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/379
2020-02-26 19:05:24 +05:30
Sebastian Dröge
35a1cedb97 qtmux: Add support for 8k resolutions in prefill mode with ProRes 2020-02-25 15:46:44 +02:00
Sebastian Dröge
3998b7cb4c rtpjitterbuffer: Include string.h for memcpy() / memset()
Usually something else is pulling it in somehow already, but not on
Windows.
2020-02-25 09:07:47 +00:00
Håvard Graff
fdf002d069 rtpsession: fix crash when no extension-header present for twcc 2020-02-24 13:06:27 +00:00
Johan Bjäreholt
ce802f033c matroska-mux: Fix incorrect rounding of timestamps
Previously we saved the buffer_timestamp straight into
mux->cluster_time. Since the cluster time saved into the file does not
have as high precision as GstClockTime depending on the timecodescale
the rounding of relative_timestamp was invalid as mux->cluster_time
which it was calculated relative to was not equal to the cluster time
written to the matroska file.

Example of "mkvinfo -v" of how it looks before and after this change in
an scenario where previously timestamps got out of order because of this
issue.

Notice the timestamp of the SimpleBlock right before and right after the
Cluster now being in order. The consequence of this however is that the
cluster timestamp is not necessarily the same as the timestamp of the
first buffer in the cluster however (in case it's rounded up).

Before

| + SimpleBlock (track number 1, 1 frame(s), timecode 126.922s = 00:02:06.922)
|  + Frame with size 432
| + SimpleBlock (track number 2, 1 frame(s), timecode 126.933s = 00:02:06.933)
|  + Frame with size 329
| + SimpleBlock (track number 2, 1 frame(s), timecode 126.955s = 00:02:06.955)
|  + Frame with size 333
|+ Cluster
| + Cluster timecode: 126.954s
| + Cluster previous size: 97344
| + SimpleBlock (key, track number 1, 1 frame(s), timecode 126.954s = 00:02:06.954)
|  + Frame with size 61239
| + SimpleBlock (track number 2, 1 frame(s), timecode 126.975s = 00:02:06.975)
|  + Frame with size 338

After

| + SimpleBlock (track number 1, 1 frame(s), timecode 135.456s = 00:02:15.456)
|  + Frame with size 2260
| + SimpleBlock (track number 2, 1 frame(s), timecode 135.468s = 00:02:15.468)
|  + Frame with size 332
| + SimpleBlock (track number 2, 1 frame(s), timecode 135.490s = 00:02:15.490)
|  + Frame with size 335
|+ Cluster
| + Cluster timecode: 135.489s
| + Cluster previous size: 158758
| + SimpleBlock (key, track number 1, 1 frame(s), timecode 135.490s = 00:02:15.490)
|  + Frame with size 88070
| + SimpleBlock (track number 2, 1 frame(s), timecode 135.511s = 00:02:15.511)
|  + Frame with size 336
2020-02-21 12:49:28 +00:00
Jake Barnes
2ffb52499f souphttpsrc: Fix cookies property
Disable session sharing and cookie jar when cookies property is set.

The cookie jar actually replaces or removes any existing Cookie header
set on the message, so the cookies property was effectively being
ignored. There doesn't appear to be a way to inject the cookies into the
jar without having to specify matching domains etc., so it's not
possible to simulate the old behaviour of unconditionally sending the
cookies with all messages, besides simply disabling the cookie jar.
2020-02-21 08:56:10 +00:00
Stefano Buora
2d3dccdba7 rtspsrc: remove useless function calls
Comparing gst_rtspsrc_loop_interleaved and gst_rtspsrc_loop_udp, and investigating on timeout issues, it sounds like a piece of code has been originally copied from udp to the interleaved one. The timeout variable is never used inside the interleaved one. No side effect has been seen in the removed function calls.

The debug message removed is pointless as the timeout used is "src->tcp_timeout" that is fixed.

The presence of the two timeout drove my team in investigating if the reference to the tcp_timeout was correct (it is). Hence we removed the misleading reference to the local timeout variable.
2020-02-20 08:27:35 +00:00
Matthew Waters
1326fcdbcc rtpbin: fix typo setting max-dropout/misorder-time
we were setting the max-dropout-time to the value of the
max-misorder-time which by default has a factor of 30 difference in
value.
2020-02-20 13:46:06 +11:00
Seungha Yang
f286f30640 qtdemux: Parse VP Codec Configuration Box
The VP Codec Configuration Box (vpcC) contains vp9 profile and
colorimetry information. Especially the profile information might
be useful for downstream to select capable decoder element.
2020-02-19 23:18:51 +09:00
Yeongjin Jeong
9feb35638a tests: flvmux: Add test for rollover timestamp
The timestamps that exceed uint32 maximum value should be handled to rollover.
2020-02-18 18:39:34 +09:00
Yeongjin Jeong
e836640bd5 flvmux: Support rollover in timestamp
For live streams, if we keep the stream for a long time, the timestamp
will be larger than max_uint32. In that case, timestamp should be handled
as a rollover timestamp rather than a backward timestamp.
2020-02-18 18:39:31 +09:00
Havard Graff
63ae338c24 rtpjitterbuffer: don't use the timer-object after JBUF_UNLOCK
It could have been freed (rtp_timer_free) in the meantime.
2020-02-17 15:04:45 +01:00
Havard Graff
1df706448c rtpmanager: Google Transport-Wide Congestion Control RTP Extension
Generating and parsing the RTCP-messages described in:
https://tools.ietf.org/html/draft-holmer-rmcat-transport-wide-cc-extensions-01
2020-02-14 10:09:02 +00:00
Håvard Graff
9ba9837058 rtpfunnel: various cleanups
* Organize GstRtpFunnelPad and GstRtpFunnel separately
* Use G_GNUC_UNUSED instead of (void) casts
* Don't call an event "caps"
* Use semicolons after GST_END_TEST (helps gst-indent)
2020-02-14 10:08:05 +00:00
Sebastian Dröge
9593a3679e qtdemux: Merge sample tables for raw audio streams with one container sample per audio sample
Instead of having chunks with one sample per raw audio sample, have
chunks with a single sample that contains lots of raw audio samples. If
necessary these are still split again later when reading the stream.

With this we are allocating a lot less memory for the parsed sample
tables and can play files that previously triggered our limit of 200MB
for the sample table. For example, one file here would previously
allocate 3.5GB for the sample table and now only allocates 70KB.
2020-02-14 08:48:01 +00:00
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