Commit graph

239 commits

Author SHA1 Message Date
Jan Schmidt
452890093d aesdec: Fix padding removal for per-buffer-padding=FALSE
When per-buffer-padding is FALSE, the OpenSSL context needs
to be told to remove any padding at the end of the ciphertext

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1243

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3406>
2022-11-15 00:13:15 +11:00
Matthew Waters
5ca3988420 webrtc/datachannel: handle error messages from appsrc/sink
Fixes a possible race where closing a data channel may produce e.g.
not-linked errors.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3381>
2022-11-11 10:13:27 +00:00
Matthew Waters
a34e380e2e sctpdec: fix stream reset (src pad removal) if no data is ever received
If we don't receive any data from usrsctp, then there will be no src pad
for the stream id and the stream reset will fail to remove the relevant
src pad.  Workaround by first attempting to add the relevant src pad, then
almost immediately removing it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3381>
2022-11-11 10:13:27 +00:00
Guillaume Desmottes
9eee5adb24 gssink: add 'content-type' property
Useful when one wants to upload a video as `video/mp4` instead of
'video/quicktime` for example.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3371>
2022-11-10 09:53:29 +00:00
Matthew Waters
e2ff6b61ce cccombiner: initial implementation of using CCBuffer helper
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3211>
2022-11-10 00:52:14 +00:00
Matthew Waters
088597b430 closedcaption: move CC buffering to helper object
Move most of the interesting code from ccconverter to this new helper
object.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3211>
2022-11-10 00:52:14 +00:00
Matthew Waters
06a20f9243 closedcaption: move cdp->cc_data into shared location
So it can be used by both ccconverter and cccombiner

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3211>
2022-11-10 00:52:14 +00:00
Matthew Waters
fde92ec43f closedcaption: move cc_data->cdp to shared file
Used by both ccconverter and cccombiner

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3211>
2022-11-10 00:52:14 +00:00
Matthew Waters
9f1b54f6ee ccconverter: avoid different indent versions indenting !! differently.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3211>
2022-11-10 00:52:14 +00:00
Matthew Waters
067185e7da closedcaption: move cdp framerate table to common file
shared by both cccombiner and ccconverter

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3211>
2022-11-10 00:52:14 +00:00
Matthew Waters
5dd199f7e8 cccombiner: don't assume a single cea608 data packet per buffer
e.g. 24fps can have up to 3 and would include either two field0 or
field1 cea608 data.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3211>
2022-11-10 00:52:14 +00:00
Matthew Waters
741cfd18b5 ccconverter: drop data when overflow on extracting cea608 from cc_data
If the buffer overflows, then drop rather than causing a failure and
fropping the output buffer indefinitely.  This may have caused downstream to
be waiting for data the will never arrive.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3211>
2022-11-10 00:52:14 +00:00
Matthew Waters
542060fea7 ccconverter: fix framerate passthrough with malformed input
If an input is malformed (only produces cea608 field 1 cc_data) then
when in passthrough we would effectively be dropping every second cea608
on output as we would not store any unused cea608 data.

Fix by having all code paths go through the framerate conversion code
which will store and retrieve any relevant data across buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3211>
2022-11-10 00:52:14 +00:00
Jan Alexander Steffens (heftig)
28628a67e5 srt: Add a property to disable automatic reconnect
This adds a new boolean property `auto-reconnect`, defaulting to `true`.

Setting it to `false` makes the elements (in caller mode) immediately
report an error to the application instead of trying to reconnect.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3326>
2022-11-07 22:23:02 +00:00
Edward Hervey
a100f36b69 webrtcbin: Don't duplicate enum string values
Some were leaked when debugging was enabled. Instead just directly use the
static strings as-is.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3347>
2022-11-07 11:21:00 +00:00
Edward Hervey
f4d0537b3e lv2: Don't leak plugin information on registration
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3319>
2022-11-04 17:59:21 +00:00
Edward Hervey
685a4aaaa7 ladspa: Don't leak plugin information on registration
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3319>
2022-11-04 17:59:21 +00:00
Edward Hervey
8ca2a2a230 fdkaacenc: Properly terminate GEnumValue table
It should be terminated with a NULL entry, otherwise we just stray into the
realms of cryptographic libraries^W^W random memory usage.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3319>
2022-11-04 17:59:21 +00:00
Jan Alexander Steffens (heftig)
424b331afc srt: Remove callers for which srt_bstats fails
This keeps them from accumulating in the element and in the stats while
the sink is not being fed, as long as we at least periodically grab
stats.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3156>
2022-11-04 13:07:34 +00:00
Jan Alexander Steffens (heftig)
d575a41145 srt: Use simpler list operations for callers
Avoid `g_list_append` and `g_list_remove` (which have to scan the list)
and replace them with `g_list_prepend` and `g_list_delete_link`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3156>
2022-11-04 13:07:34 +00:00
Jan Alexander Steffens (heftig)
3c23c16f40 srt: Clean up poll/sock lifecycle
Make sure `srtobject->poll_id` is never invalid as long as `srtobject`
exists. Only remove our caller socket from it when the socket becomes
invalid.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3156>
2022-11-04 13:07:34 +00:00
Jan Alexander Steffens (heftig)
4e05100e8c srt: Clean up error handling
- Make the srt_epoll_wait loops more uniform.

- Error only via GError when possible; let the element send the error
  message. Avoids a second error message.

- Return 0 when cancelled. Avoids an error message from the element.

- Don't send an error message from send_headers when we're a server
  sink.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3156>
2022-11-04 13:07:34 +00:00
Jan Alexander Steffens (heftig)
a3cc5cf257 srt: Simplify socket stats
Don't hide stats depending on whether we're a sending or receiving
socket. While we're here, add some more debug logs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3156>
2022-11-04 13:07:34 +00:00
Jan Alexander Steffens (heftig)
b6974b6afc srt: Replace stats accumulation with naive byte counting
srt_bstats cannot be used to get the stats of closed connections, so the
best we can do is keep the running count ourselves.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3156>
2022-11-04 13:07:34 +00:00
Guillaume Desmottes
a92f41e0c7 wpe: fix wpevideosrc gst-play example
wpe:// no longer works since 1.20, see wpesrc examples.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3306>
2022-11-02 00:21:21 +00:00
Sanchayan Maity
da52bedbff fdkaacenc: Update documentation to clarify bitrate and peak-bitrate
bitrate property is only applicable for constant bitrate and
peak-bitrate is only applicable for variable bitrate. Clarify
the same.
2022-10-30 16:54:51 +05:30
Sanchayan Maity
f0ceb9ea4f fdkaacenc: Add support for setting bitrate mode 2022-10-30 16:54:51 +05:30
Sanchayan Maity
595dd7a1ed fdkaacenc: Add support for setting peak bitrate 2022-10-29 16:04:42 +05:30
Sanchayan Maity
734593ccab fdkaacenc: Add support for enabling afterburner
This is an additional quality parameter. In the default configuration this
quality switch is deactivated because it would cause a workload increase
which might be significant. If workload is not an issue in the application
it can be recommended to activate this feature.
2022-10-29 15:57:52 +05:30
Sanchayan Maity
a63d8ee720 fdkaacdec: Do not report decoding error for flush request
A flush request is done when set_format is called to empty internal bit
buffer maintained by fdk-aac. When this happens, during the explicit
call to handle_buffer, decodeFrame does not return a AAC_DEC_OK. This
gets reported as a decoding error while no decoding error in fact took
place. Since this can be confusing, just return a GST_FLOW_OK and log
that an explicit flush was requested.
2022-10-29 10:47:16 +05:30
Thibault Saunier
4f991a55af adaptivedemux: Minor typo fix
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3159>
2022-10-27 19:45:44 +00:00
Thibault Saunier
8a9821e805 dash: Fix computing repeat_index when seeking in stream with a start !=0 on the first fragment
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3159>
2022-10-27 19:45:44 +00:00
Tim-Philipp Müller
d7e2aff994 fdkaacenc: fix output caps in case of implicit signaling and HE-AAC
Need to put the actual profile in the output caps otherwise any
capsfilter after the encoder that was used to force the output
profile will fail, such as

  fdkaacenc ! audio/mpeg,stream-format=adts,profile=he-aac-v1 ! ..

because we put profile=lc in there to match the profile signaled
in the ADTS header. This is expressed through the base-profile=lc
in the GStreamer caps though, the profile needs to carry the
'real' profile.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1785>
2022-10-25 00:13:04 +00:00
Tim-Philipp Müller
24645e35c5 fdkaacenc: don't set base-profile=lc for non-backwards compatible output
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1785>
2022-10-25 00:13:04 +00:00
Tim-Philipp Müller
31c04f87e3 fdkaacenc: rename profile=sbr|ps to profile=he-aac-v1|he-aac-v2
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1785>
2022-10-25 00:13:04 +00:00
Piotrek Brzeziński
d8b1ff4668 fdkaacenc: add support for AAC-LD
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1785>
2022-10-25 00:13:04 +00:00
Piotrek Brzeziński
8cda666cb0 fdkaacenc: add support for HE-AACv1 and HE-AACv2
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1785>
2022-10-25 00:13:04 +00:00
Matthew Waters
0077d13304 webrtcbin: configure rtpulpfecdec passthrough property
This allows downstream (payloaders mostly) to be able to correctly
detect actual packet loss from rtp sequence numbers.

See
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/581
for background.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1407

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3212>
2022-10-23 23:44:07 +00:00
Matthew Waters
a633f5d287 webrtcbin: also add rtcp-fb ccm fir for video mlines by default
In addition to the 'nack pli' already added.  Both are supported by
rtpbin/rtpsession by default already.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3235>
2022-10-21 01:02:34 +00:00
Sangchul Lee
0f05be382b webrtcbin: Improve documentation of 'turn-server' property
Description about how to set time-limited credentials is added.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3229>
2022-10-20 15:30:07 +00:00
Fabian Orccon
50c6c54675 srtp: Fix test skipping when plugin option is disabled
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3200>
2022-10-18 22:12:41 +00:00
Johan Sternerup
44eea7bd8a sctpenc: Prohibit sending of interleaved message parts
Apparently we cannot start sending messages from another datachannel
before the previous message was completely sent. usrsctplib will
complain about being locked on another stream id and set
errno=EINVAL.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2454>
2022-10-11 09:36:13 +00:00
Xavier Claessens
56eb44c502 Meson: Fix libxml2 fallback
The variable xml2lib_dep does not exist. The correct name is already in
the wrap file.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3136>
2022-10-07 07:56:21 -04:00
Sangchul Lee
93b896eb4e webrtcbin: Fix pointer dereference before null check
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3129>
2022-10-06 16:46:33 +00:00
Johan Sternerup
212c09a70e webrtc: return error when sending on non-open datachannel
According to W3C
specification (https://w3c.github.io/webrtc-pc/#datachannel-send) we
should return InvalidStateError exception when trying to send when the
channel is not open. In the world of C/glib/gstreamer we don't have
exceptions but have to rely on gboolean/GError instead. Introducing
these calls for a change in function signature of the action signals
used to send data on the datachannel. Changing the signature of the
existing "send-string" and "send-data" signals would mean an immediate
breaking change so instead we deprecate them. Furthermore, there is no
way to express GError** as an argument to an action signal in a way
that fits language bindings (pointer-to-pointer simply does not work)
and we have to use regular functions instead.

Therefore we introduce gst_webrtc_data_channel_send_data_full() and
gst_webrtc_data_channel_send_string_full() while deprecating the old
functions and corresponding signals.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1958>
2022-10-05 11:08:30 +00:00
Devin Anderson
31831eb47e voamrwbenc: Fix truncation of audio data at end-of-stream when audio data
doesn't align on 20 millisecond frame size.

The AMR-WB codec imposes a fixed 20 millisecond frame size.  In its current
form, the `voamrwbenc` plugin deals with this limitation by discarding any
audio at the end of the stream that falls short of 20 milliseconds.  This patch
keeps the audio data, and appends silence to the end to preserve frame size
alignment.

The patch also adds tests to check for the updated behavior.  I noticed that
tests weren't being built, so I changed the build to allow for building the
tests when the `tests` and `voamrwbenc` options are set.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3027>
2022-09-16 00:14:58 +00:00
Mathieu Duponchelle
b454ec972f webrtcbin: fix picking available payload types
When picking an available payload type, we need to pick one that is
available across all media.

The previous code, when multiple media were present, looked at the first one,
noticed it had pt 96 as the media pt, then simply looked at the next media,
noticed it didn't, and decided 96 was available.

Instead, check if the pt is used by any of the media, if it is, decide
it is not available and go to the next pt. I'm fairly sure that was the
original intent.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2984>
2022-09-07 03:22:34 +00:00
Jordan Petridis
a7f9c97454 fluidsynth: correctly version guard methods
We bumped the minimum version to 2.1 but the api we used
wasn't introduced till version 2.2 of fluidsynth

Follow-up to gstreamer/gstreamer!2718

https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2718

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2835>
2022-09-05 17:48:27 +00:00
Jan Schmidt
4e25c519de dashdemux: Preserve current representation on live manifest updates
When updating a manifest during live playback, preserve the current
representation for each stream.

During update_fragment_info, if the current representation changed
because it couldn't be matched, trigger a caps change and new
header download.

This reverts commit e0e1db212f
and reapplies "dashdemux: Fix issue when manifest update sets slow start
without passing necessary header & caps changes downstream" with
changes.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/507
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1729

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2920>
2022-09-05 16:07:00 +00:00
Olivier Crête
4b3b234f72 webrtcbin: Allow locked mlines with no caps, as the last ones
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2439>
2022-09-02 11:52:58 +02:00