Commit graph

290 commits

Author SHA1 Message Date
Guillaume Desmottes
3d1390d31a rtpptdemux: set different stream-id on each src pad
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3855>
2023-02-01 09:17:33 +00:00
Guillaume Desmottes
cc2b8f6ae8 rtpssrcdemux: set different stream-id on each src pad
All the RTP src pads were sharing the same stream-id while each actually
carry a different stream.

This was causing problem for example when funneling the streams together
and then trying to split them using 'streamiddemux'.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3855>
2023-02-01 09:17:33 +00:00
Sebastian Dröge
3ca85189fd rtspsrc: Also consider "Method Not Valid In This State" error in broken control URL handling workaround
Some servers send a 455 error instead of any reasonable error when using
a correctly constructed control URL.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3854>
2023-02-01 07:55:24 +00:00
Alicia Boya García
8a6023a38a qtdemux: Use safer clearing functions in dispose()
In theory, `dispose()` functions should be idempotent and should be
prepared not to crash or cause a double-free if an unref done from
inside caused a recursive call to `dispose()` of the same object.

https://developer.gnome.org/gobject/stable/howto-gobject-destruction.html

This patch modifies the `dispose()` method to honor these constraints.

Since the double `dispose()` call won't actually occur in qtdemux (there
is no cycle detection mechanism that could invoke it to work that way),
this is more of a code cleanup than a user-facing problem fix.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3822>
2023-01-28 00:32:57 +00:00
Daniel Knobe
5e9a32ed8c imagefreeze: add bayer support
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3807>
2023-01-26 21:30:51 +00:00
Mathieu Duponchelle
2048a0a4d9 redenc: fix setting of extension ID for twcc
1 was previously hardcoded in, and the bug went under the radar because
webrtcsink hardcodes the number too.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3785>
2023-01-24 22:52:48 +00:00
Tim-Philipp Müller
74e103e53f xingmux: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
2023-01-24 15:25:06 +00:00
Tim-Philipp Müller
fc82621e09 multiudpsink: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
2023-01-24 15:25:06 +00:00
Tim-Philipp Müller
8222b97331 rtpmanager: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
2023-01-24 15:25:06 +00:00
Tim-Philipp Müller
e66f8cff26 rtp: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
2023-01-24 15:25:06 +00:00
Tim-Philipp Müller
56d3beed0b multifile: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
2023-01-24 15:25:06 +00:00
Tim-Philipp Müller
e256472ca6 matroska: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
2023-01-24 15:25:06 +00:00
Tim-Philipp Müller
172c6ca1dc flv: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
2023-01-24 15:25:06 +00:00
Tim-Philipp Müller
9f4c514c52 dtmf: drop use of GSlice
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3695>
2023-01-24 15:25:06 +00:00
David Svensson Fors
d0edc1ad6a udpsrc: GstSocketTimestampMessage only for SCM_TIMESTAMPNS
Deserialize socket control messages as GstSocketTimestampMessage only
if (level, type) is (SOL_SOCKET, SCM_TIMESTAMPNS).

Without this patch, messages with types SCM_RIGHTS or SCM_CREDENTIALS
could be deserialized as GstSocketTimestampMessage instead of
GUnixFDMessage or GUnixCredentialsMessage from gio.

Fixes 

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3777>
2023-01-24 10:49:01 +01:00
Hiero32
145d362129 taginject: Add scope property
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3697>
2023-01-24 00:20:53 +00:00
Sebastian Dröge
067b5d92b4 matroska: Add stream-format = (string) obu-stream to AV1 caps
Anything else is not allowed in Matroska/WebM.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3740>
2023-01-19 12:10:40 +02:00
Sebastian Dröge
4c8141a0c3 isomp4: Add stream-format = (string) obu-stream to AV1 caps
Anything else is not allowed in MP4.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3740>
2023-01-19 12:10:40 +02:00
Jan Alexander Steffens (heftig)
211191564e qtdemux: Add basic support for AVC-Intra video
AVC-Intra is a range of H.264-compliant intra-only codecs from
Panasonic. The codes and descriptions have been taken from VLC.

The (encumbered) sample I have here produces byte-stream H.264,
including SPS and PPS and no `avcC` box.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3739>
2023-01-18 10:01:30 +00:00
Olivier Crête
c593930055 rtopuspay: Use GstStaticCaps to cache parsed caps
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3674>
2023-01-12 18:48:35 -05:00
Olivier Crête
46a6f72f03 rtopuspay: Ignore the stereo parameter in multiopus caps
Also add unit tests for the various variants

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3674>
2023-01-12 18:48:35 -05:00
Olivier Crête
f1cf457811 rtpopuspay: Leave original caps as-is
This should make it work if someone specifies stereo with MULTIOPUS
somehow.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3674>
2023-01-12 18:48:35 -05:00
Olivier Crête
c52c66b575 rtpopuspay: Return upstream channel filter based on OPUS vs MULTICAPS
Only allow 1 or 2 channels if the caps are OPUS, or 3+ if they are
MULTIOPUS.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3674>
2023-01-12 18:48:35 -05:00
Olivier Crête
c51ae6112d rtpopus: Put MULTIOPUS in all caps
The RTP payload encoding-name are always in caps in GStreamer.
In SDP, they are not case-sensitive, but since caps are, we need to pick
a caps, and we picked upper-case along time ago.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3674>
2023-01-12 18:48:35 -05:00
Seungha Yang
6540c4e89c rtspsrc: Fix string leak
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3645>
2022-12-28 04:39:18 +09:00
Seungha Yang
9b305df1cc rtptimerqueue: Fix memory leak
Should chain up to parent's finalize

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3645>
2022-12-27 19:31:16 +00:00
Patricia Muscalu
d752bf1b46 qtmux: Fix buffer leak in fragment_buffers
When pushing buffers from one of the sink pads fail,
make sure that all buffers added to fragment_buffers on other pads
are freed as well.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3624>
2022-12-22 14:11:10 +00:00
Mathieu Duponchelle
194dcd91e0 qtmux: For video with N/1001 framerates use N as timescale instead of centiframes
This is recommended by various specifications for such framerates, while
for integer framerates we continue using centiframes to allow for some
more accuracy.

Using N means that no rounding error accumulates, eventually leading to
outputting a packet with a different duration.

Some tools such as MediaInfo determine that a stream is variable
framerate if any packet has a different duration than the others, and
there is no reason I can see for not using the full 4 bytes of
resolution that the mp4 timescale offers.

Example problematic pipeline:

```
videotestsrc num-buffers=5001 ! video/x-raw,framerate=60000/1001,width=320,height=240 ! \
videoconvert ! x264enc bitrate=80000 speed-preset=1 tune=zerolatency ! h264parse ! \
video/x-h264,profile=high-10 ! mp4mux ! filesink location="result2.mp4"
```

This results in a media file that MediaInfo detects as variable
framerate because the 5000th packet has duration 99 instead of 100.

With this patch, the timescale is 60000 and all packets have duration
1001.

Related issue for context: https://bugzilla.gnome.org/show_bug.cgi?id=769041

Co-authored-by: Sebastian Dröge <sebastian@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3049>
2022-12-22 12:31:06 +02:00
Sebastian Dröge
066558cba1 qtdemux: Always use tfdt if available in BYTE segments
This reverts the decision from
  https://bugzilla.gnome.org/show_bug.cgi?id=754230
where it was decided that we rather play safe and only use the `tfdt` if
it is "significantly different" to the sum of sample durations.

As the specification says

    If the time expressed in the track fragment decode time (‘tfdt’) box
    exceeds the sum of the durations of the samples in the preceding
    movie and movie fragments, then the duration of the last sample
    preceding this track fragment is extended such that the sum now
    equals the time given in this box.

we have to use the `tfdt` in general to allow for it to signal gaps in
the stream.

A muxer producing fragments might not yet know the full duration of the
last sample of a previous fragment if the next fragment starts with a
gap, and knowing the actual start of the next fragment would potentially
require to violate latency requirements.

Additionally, the existence of `tfdt` allows to avoid accumulating
rounding errors from summing up the durations.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3586>
2022-12-17 19:26:19 +02:00
Xabier Rodriguez Calvar
87ae60176b qtdemux: Clear protection events when we get new ones
If we keep the old events they can be end up being passed to the app, that could
discard the protection information because it has been seen before.

Drive by improvement: use g_queue_clear_full instead of foreach+clear for
protection events.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3547>
2022-12-14 11:01:23 +01:00
Mathieu Duponchelle
fa71217502 rtpvp9depay: expose keyframe-related properties
This simply brings in the wait-for-keyframe and request-keyframe
properties from rtpvp8depay.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/909>
2022-12-10 13:28:07 +00:00
Jacek Skiba
61c17c5665 qtdemux: exit when protection caps are not defined during PIFF parsing
Reproduction testcase (uses PlayReady):
https://developers.canal-plus.com/rx-player/upc/?appTileLocation=[object%20Object]

In test streams we are using PIFF box, but caps did not had
present GST_PROTECTION_SYSTEM_ID_CAPS_FIELD. In consequence, invalid
system_id was returned which caused SIGSEGV crash.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3535>
2022-12-07 18:35:37 +00:00
Philippe Normand
b9011f3541 flacparse: Fix handling of headers advertising 32bps
According to the flac bitstream format specification, the sample size in bits
corresponding to `111` is 32 bits per sample.

https://xiph.org/flac/format.html#frame_header

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3517>
2022-12-04 11:47:57 +00:00
Aleksandr Slobodeniuk
38f6a0ba2e rtspsrc: fix seek event leaks
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3500>
2022-12-01 23:52:40 +00:00
Matt Crane
ca7f66f9b5 rtpsession: Support disabling late adjustment of ntp-64 header ext
Currently in rtp_session_send_rtp(), the existing ntp-64 RTP header
extension timestamp is updated with the actual NTP time before sending
the packet. However, there are some circumstances where we would like
to preserve the original timestamp obtained from reference timestamp
buffer metadata.

This commit provides the ability to configure whether or not to update
the ntp-64 header extension timestamp with the actual NTP time via the
update-ntp64-header-ext boolean property. The property is also exposed
via rtpbin. Default property value of TRUE will preserve existing
behavior (update ntp-64 header ext with actual NTP time).

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1580

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3451>
2022-11-24 08:23:03 +00:00
Johan Sternerup
9794c9bfd0 Use the correct SSRC(s) when routing a RTCB FB FIR
Previously we tried to route an incoming RTCP FB FIR to the correct ssrc
using the "media source" component of the RTCP FB message. However,
according to RFC5104 (section 4.3.1.2) the "media source" SHALL be set
to 0. Instead the ssrc(s) in use are propagated via the FCI data. Now
a specific GstForceKeyUnit event is sent for every ssrc.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3292>
2022-11-23 11:31:23 +00:00
Jan Schmidt
cb225b3682 rtpsource: Track the seqnum for senders
RTP source statistics are tracked for local senders by
treating them as a receiver of their own outbound packets.

Accordingly, track the highest packet seqnum so that the
packets-lost calculation generates a sensible number instead
of always reporting -$number_of_packets_sent

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3454>
2022-11-23 10:26:29 +00:00
Jan Alexander Steffens (heftig)
1d7c936db0 rtspsrc: Don't replace 404 errors with "no auth protocol found"
When getting a "404 Not Found" response from the DESCRIBE request, the
source produced a "No supported authentication protocol was found" error
instead of passing on the 404, which was confusing.

Only produce this error message when we're handling a response of "401
Unauthorized" without a compatible WWW-Authenticate header.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3414>
2022-11-22 13:07:17 +00:00
Edward Hervey
f3c2f612ce rtspsrc: Don't leak sticky events
We have incremented the reference 2 lines above, and
gst_pad_store_sticky_event() does not take a reference, therefore release it

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3443>
2022-11-21 19:02:44 +00:00
Vivia Nikolaidou
f29c19be58 splitmuxsink: Avoid assertion when WAITING_GOP_COLLECT on reference context
I have seen a backtrace out in the wild where this happened. Maybe after
receiving EOS and stream-start on the reference context.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3005>
2022-11-18 15:52:03 +00:00
Edward Hervey
845dcf7ec5 imagesequencesrc: Don't leak caps
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3428>
2022-11-18 07:22:23 +00:00
Matthew Waters
8e355d23a1 qtmux: use trun with multiple entries in more cases
The only case where we definitely need to write a new trun is when the
data_offset value does not match the end of the list of entries.
Needing multiple trun atoms is required when interleaving multiple
streams together.

All other cases can be covered by adding more entries to the existing
trun atom.

Fixes playback of fragemented mp4 in ffplay and chrome.

Using e.g. mp4mux fragment-duration=1000 fragment-mode=dash-or-mss
and
mp4mux fragment-duration=1000 fragment-mode=first-moov-then-finalise

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3426>
2022-11-17 21:04:57 +11:00
Nirbheek Chauhan
13723198a1 rtspsrc: Fix regression when using hostname in the location property
When the address can't be parsed as an IP address, it should just be
treated as a hostname and used as-is.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1576

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3420>
2022-11-16 11:30:26 +00:00
Sebastian Dröge
3d79402344 rtpjitterbuffer: Reschedule timers when updating their offset
As EXPECTED timers are skipped the order of the timers relative to each
other can change if there are EXPECTED timers and rescheduling needs to
happen.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1422

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3416>
2022-11-16 08:26:41 +00:00
Sanchayan Maity
02fd7fb777 wavparse: Do not run all typefinders for all output
In order to figure out if the "raw" audio contained within the wav
container is actually DTS, wavparse calls the typefinder helper
except that means it runs all typefinders.

Since it only cares about checking for DTS, we should only run the
audio/x-dts typefinder (if present). Commit 858e516 did not really
fix things.

Use the new type helper with the caps to fix this.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3417>
2022-11-16 10:32:25 +05:30
Sebastian Dröge
424e208170 rtspsrc: Consistently set seqnums on events
Set udpsrc seqnums on all events sent to the udpsrc's, and before
forwarding events out of rtspsrc set the latest seek seqnum on them if
any.

Also produce a consistent seqnum in rtspsrc from the very beginning.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3409>
2022-11-16 02:36:30 +00:00
Sebastian Dröge
e6efd288c2 rtspsrc: Make segment event writable before overriding the seqnum and use the proper API to do so
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3409>
2022-11-16 02:36:30 +00:00
Sebastian Dröge
4099fd064b rtspsrc: Intercept and handle events when using no manager too
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3409>
2022-11-16 02:36:30 +00:00
Sebastian Dröge
e6a2e41c06 rtspsrc: Don't blindly copy over sticky events from manager pad to external source pad
This would get around the code that modifies some events when they go
through the ghost pad's proxypad. Instead go via the event function.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3409>
2022-11-16 02:36:30 +00:00
Sebastian Dröge
a4674a1e17 rtspsrc: Don't make udpsrc segment events writable just to retrieve their seqnum
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3409>
2022-11-16 02:36:30 +00:00