Commit graph

327 commits

Author SHA1 Message Date
Sebastian Dröge
2bd21a6f40 rtpjitterbuffer: Avoid integer overflow in max saveable packets calculation with negative offset
The timestamp offset can be negative, and it can be a bigger negative
number than the latency introduced by the rtpjitterbuffer so the overall
timeout offset can be negative.

Using the negative offset for calculating how many packets can still
arrive in time when encountering a lost packet in an equidistant stream
would then overflow and instead of considering fewer packets lost a lot
more packets are considered lost.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5318>
2023-09-12 10:40:33 +01:00
Alicia Boya García
654ca283a7 qtdemux: Fix premature EOS when some files are played in push mode
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2771

This EOS branch exists so that if a seek with a stop is made, qtdemux
stops accepting bytes from the sink after the entire requested playback
range is demuxed, as otherwise we could keep download content that is
not being used.

This patch fixes two flaws that were present in that EOS check:

1) A comparison was made between track time and movie time without conversion.
This made the check trigger early in files with edit lists. This patch fixes
this by converting the track PTS to movie PTS (stream time) for the check.

2) To avoid sending a EOS prematurely when the segment stop is within a GOP and
B-frames are present, the check for EOS should only be done for keyframes. I
gather this was already the intention with the existing code, but because it
used `stream->on_keyframe` instead of the local variable `keyframe` the old
code was checking if the *previous* frame was a keyframe.

It's interesting to note that these two flaws in the old code mask each other
in most cases: the track PTS will have reached the movie end PTS, but EOS would
only be sent if the previous frame was a keyframe. A simple case where they
wouldn't mask each other, reproducing the bug, is a sequence of 3 frame GOPs
with structure I-B-P.

The following validateflow tests have been added to future-proof the
fix:

 * validate.test.mp4.qtdemux_ibpibp_non_frag_pull.default
 * validate.test.mp4.qtdemux_ibpibp_non_frag_push.default

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5114>
2023-08-03 20:22:51 +00:00
Guillaume Desmottes
bbca6cc8c8 videoflip: fix concurrent access when modifying the tag list
We were checking if the tag list is writable, but it may actually be
shared through the same event (tee upstream or multiple consumers).

Fix a bug where multiple branches have a videoflip element checking the
taglist. The first one was changing the orientation back to rotate-0
which was resetting the other instances.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5099>
2023-07-25 17:32:50 +01:00
Xabier Rodriguez Calvar
1a3deef85f qtdemux: attach cbcs crypt info at the right moment
Before it was always added but that can cause issues when the stream begins
unencrypted.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5096>
2023-07-25 12:43:28 +01:00
David Craven
ca48193cd6 matroska: demux: Strip signal byte from encrypted blocks
Removes the signal byte when the frame is unencrypted to
be consistent with when the frame is encrypted.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5014>
2023-07-11 13:53:02 +00:00
Guillaume Desmottes
4c77bc2161 videoflip: fix critical when tag list is not writable
Fix this pipeline where the tag list is not writable:

gst-launch-1.0 videotestsrc ! taginject tags="image-orientation=rotate-90" ! videoflip video-direction=auto \
  ! autovideosink

GStreamer-CRITICAL **: 12:34:36.310: gst_tag_list_add: assertion 'gst_tag_list_is_writable (list)' failed

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4990>
2023-07-07 14:59:33 +01:00
Seungha Yang
4bd7992d14 rtspsrc: Fix crash when is-live=false
The pad's parent (i.e., rtspsrc) can be nullptr since we add pads
later.

Co-authored-by: Jan Schmidt <jan@centricular.com>

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2751
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4971>
2023-07-05 09:17:14 +00:00
Edward Hervey
bf0446ad73 matroska-demux: Properly handle early time-based segments
Refusing an incoming segment in < GST_MATROSKA_READ_STATE_DATA should only be
done if the incoming segment is not in GST_FORMAT_TIME.

In GST_FORMAT_TIME, we are just storing the values and returning, so we can
invert the order of the checks.

Fixes proper segment propagation in matroska/webm DASH use-cases

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4922>
2023-06-23 08:55:47 +02:00
Guillaume Desmottes
2dc0e1ac87 videoflip: update orientation tag in auto mode
The frames are flipped according to the tag orientation so it's no longer accurate.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4796>
2023-06-22 14:06:33 +02:00
François Laignel
107c456c0d qtdemux: opus: set entry as sampled
... otherwise streams with constant size samples defined with a single
`sample_size` for all samples in the `stsz` box fall in the category
`chunks_are_samples` in `qtdemux_stbl_init`, overriding the actual
sample count.

`FOURCC_soun` would set this automatically for `compression_id == 0xfffe`,
however `compression_id` is read from the Audio Sample Entry box at an offset
marked as "pre-defined" in some version of the spec and set to 0 both by
GStreamer and FFmpeg for opus streams.

Considering the stream `sampled` flag is set explicitely by other fourcc
variants, doing so for opus seems consistent.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4908>
2023-06-20 20:35:49 +01:00
Sebastian Dröge
bdc8021c73 flacparse: Avoid integer overflow in available data check for image tags
If the image length as stored in the file is some bogus integer then
adding it to the current byte readers position can overflow and wrongly
have the check for enough available data succeed.

This then later can cause NULL pointer dereferences or out of bounds
reads/writes when actually reading the image data.

Fixes ZDI-CAN-20775
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2661

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4897>
2023-06-20 09:16:37 +01:00
François Laignel
670ce68f06 qtdemux: parse Opus and dOps as qtdemux nodes and add size checks
This allows checking the nodes conformity and dumping parsed values.

Note: Audio Sample Entry version parsing and offset handling is handled as part
of `FOURCC_soun` common processing and in `qtdemux_parse_node`.

Also, only read `stream_count` and `coupled_count` when
`channel_mapping_family` != 0. See:

https://opus-codec.org/docs/opus_in_isobmff.html#4.3.2

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4891>
2023-06-19 16:09:48 +01:00
François Laignel
8d2dc95567 qtdemux: fix byte order for opus extension and version field type
The "Encapsulation of Opus in ISO Base Media File Format" [1] specifications,
§ 4.3.2 Opus Specific Box, indicates that data must be stored as big-endian.

[1] https://opus-codec.org/docs/opus_in_isobmff.html#4.3.2

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4891>
2023-06-19 16:09:48 +01:00
François Laignel
9e27d36edc qtmux: fix byte order for opus extension
The "Encapsulation of Opus in ISO Base Media File Format" [1] specifications,
§ 4.3.2 Opus Specific Box, indicates that data must be stored as big-endian.

In `build_opus_extension`, `gst_byte_writer_put*_le ()` variants were used,
causing audio streams conversion to Opus in mp4 to offset samples due to the
PreSkip field incorrect value (29ms early in our test cases).

[1] https://opus-codec.org/docs/opus_in_isobmff.html#4.3.2

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4891>
2023-06-19 16:09:48 +01:00
Jan Alexander Steffens (heftig)
21cccd0e00 isomp4: Fix (E)AC-3 channel count handling
The muxer used a fixed value of 2 channels because the TR 102 366 spec
says they're to be ignored. However, the demuxer still trusted them,
resulting in bad caps.

Make the muxer fill in the correct channel count anyway (FFmpeg already
does) and make the demuxer ignore the value.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4773>
2023-06-15 15:23:22 +00:00
Jochen Henneberg
33e789e067 rtspsrc: Cleanup code for next pending command
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4799>
2023-06-08 07:22:59 +02:00
Jochen Henneberg
89ece711dd rtspsrc: Do not try send dropped get/set parameter
If the set_get_param_q has been emptied we have to reset the cached
pending command to CMD_LOOP as we will not have the request parameters
anymore.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4799>
2023-06-08 07:22:59 +02:00
Michael Olbrich
9f169ffaf9 flvmux: use the correct timestamp to calculate wait times
Since c0bf793c05 ("flvmux: Set PTS based on
running time") the timestamp of the output buffer is already in running
time. So using that for 'srcpad->segment.position' does not work correctly
because gst_aggregator_simple_get_next_time() will convert it again with
gst_segment_to_running_time().
This means that the timestamp returned by
gst_aggregator_simple_get_next_time() may be incorrect. For example, if
flvmux is added to a already runinng pipeline then the timestamp is too
small and gst_aggregator_wait_and_check() returns immediately. As a result,
buffers may be muxed in the wrong order.

To fix this, use the PTS of the incoming buffer instead of the outgoing
buffer. Also add the duration as get_next_time() is supposed to return the
timestamp of the next buffer, not the current one.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4734>
2023-05-30 00:52:01 +01:00
Jan Schmidt
cd20649450 splitmuxsrc: Make PTS contiguous by preference
Make splitmuxsrc deal better with stream reordering by
making the largest observed PTS contiguous in the
next fragment. Previously, it selected DTS, but then
aligned that with the segment start of the next fragment,
which holds PTS values - leading to glitches in
streams that don't have PTS = DTS at the start.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4660>
2023-05-17 17:26:20 +00:00
Sebastian Dröge
59637a244c qtmux: Fix extraction of CEA608 data from S334-1A packets
The index is already incremented by 3 every iteration so multiplying it
by 3 additionally on each array access is doing it twice and does not
work.

This caused invalid files to be created if there's more than one CEA608
triplet in a buffer, and out of bounds memory reads.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4645>
2023-05-16 14:59:45 +01:00
Mathieu Duponchelle
8bb2d23666 Revert "qtdemux: fix conditions for end of segment in reverse playback"
This reverts commit 9deb3c27ac.

The test case that was described in the associated MR
(https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/262)
remains adequately fixed by a related MR that was merged later
(https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/275).

It introduced incorrect logic that broke edit lists as described in
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2549

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2549
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4605>
2023-05-11 18:44:15 +01:00
François Laignel
b9f7ab6052 rtpmanager/rtsession: data race leading to critical warnings
This is a fix for a data race leading to:

> GLib-CRITICAL: g_hash_table_foreach:
>   assertion 'version == hash_table->version' failed

Identified sequence:

* `rtp_session_on_timeout` acquires the lock on `session` and proceeds with its
  processing.
* `rtp_session_process_rtcp` is called (debug log : received RTCP packet) and
  attempts to acquire the lock on `session`, which is still held by
  `rtp_session_on_timeout`.
* as part of an hash table iterator, `rtp_session_on_timeout` transitively
  invokes `source_caps` which releases the lock on `session` so as to call
  `session->callbacks.caps`.
* Since `rtp_session_process_rtcp` was waiting for the lock to be released, it
  succeeds in acquiring it and proceeds with `rtp_session_process_rr` which
  transitively calls `g_hash_table_insert` via `add_source`.
* After `source_caps` re-acquires the lock and gives the control flow back to
  `rtp_session_on_timeout`, the hash table iterator is changed, resulting in the
  assertion failure.

This commits copies `sess->ssrcs[sess->mask_idx]` and iterates on the copy so
the iterator is not affected by a concurrent change due to the lock being
released in the `source_caps` callback.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4585>
2023-05-09 22:35:23 +00:00
Philippe Normand
9f8d69540c rtpdtmfdepay: Classify as RTP element
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4584>
2023-05-09 17:09:22 +01:00
Philippe Normand
ff271e1741 rtpdtmfsrc: Classify as RTP source
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4584>
2023-05-09 17:09:22 +01:00
Xabier Rodriguez Calvar
5c863418ba qtdemux: emit no-more-pads after pruning old pads
If we don't do that, clients can rely on this signal to see the final pad
topology but it won't be the real one as some of them will disappear after
emitting that signal. This can happen after injecting a different init segment.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4557>
2023-05-05 16:24:17 +01:00
Mathieu Duponchelle
b17fbb231c videoflip: fix setting of method property at construction time
Since c2f890ab, element properties are gathered from the parse-launch
line and passed at object construction.

This caused the following issue to happen in videoflip:

* videoflip installed a CONSTRUCT property named method, now deprecated
* videoflip now also overrides that property with a video-direction
  property

GObject construction causes method to be set first at construct time,
with the user-provided value, then video-direction with the default
value.

The user-provided value was thus overridden, causing a regression.

Fix by not installing the properties as CONSTRUCT, and explicitly
implementing constructed() instead in order to ensure that we do still
call gst_video_flip_set_method() at least once during construction.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2529

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4551>
2023-05-05 11:58:37 +01:00
François Laignel
943a53cc51 rtpmanager/rtsession: race conditions leading to critical warnings
While testing the [implementation for insertable streams] in `webrtcsink` &
`webrtcsrc`, I encountered critical warnings, which turned out to result from
two race conditions in `rtpsession`. Both race conditions produce:

> GLib-CRITICAL: g_hash_table_foreach:
>   assertion 'version == hash_table->version' failed

This commit fixes one of the race conditions observed.

In its simplest form, the test consists in 2 pipelines and a Signalling server:

* pipelines_sink: audiotestsrc ! webrtcsink
* pipelines_src: webrtcsrc ! appsrc

1. Set `pipelines_sink` to `Playing`.
2. The Signalling server delivers the `producer_id`.
3. Initialize `pipelines_src` to establish a session with `producer_id`.
4. Set `pipelines_src` to `Playing`.
5. Wait for a buffer to be received by the `appsrc`.
6. Set `pipelines_src` to `Null`.
7. Set `pipelines_sink` to `Null`.

The race condition happens in the following sequence:

* `webrtcsink` runs a task to periodically retrieve statistics from `webrtcbin`.
  This transitively ends up executing `rtp_session_create_stats`.
* `pipelines_sink` is set to `Null`.
* In `Paused` to `Ready`, `gst_rtp_session_change_state()` calls
  `rtp_session_reset()`.
* The assertion failure occurs when `rtp_session_reset` is called while
  `rtp_session_create_stats` is executing.

This is because `rtp_session_create_stats` acquires the lock on `session` prior
to calling `g_hash_table_foreach`, but `rtp_session_reset` doesn't acquire the
lock before calling `g_hash_table_remove_all`.

Acquiring the lock in `rtp_session_reset` fixes the issue.

[implementing insertable streams support]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1176

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4532>
2023-05-03 09:59:22 +01:00
Sebastian Dröge
9e2eeab1c6 Revert "splitmuxsink: Avoid assertion when WAITING_GOP_COLLECT on reference context"
This reverts commit f29c19be58. If this is
called for the reference context then we would run into an infinite
loop, which is not really better than an assertion.

By fixing up DTS to never be ahead of the PTS in the previous commit
this situation should be impossible to hit now.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4515>
2023-04-30 15:16:00 +01:00
Sebastian Dröge
35322de964 splitmuxsink: Catch invalid DTS to avoid running into problems later
DTS > PTS makes no sense, so we clamp DTS to the PTS. Also if there's a
PTS but no DTS, then assume that PTS=DTS to make sure we're not working
with a much older DTS.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4515>
2023-04-30 15:16:00 +01:00
Sebastian Dröge
5c4a356164 rtspsrc: Fix handling of * control path
Regression introduced by 7f9d689572.
Thanks to Tristan Matthews for reporting this.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4503>
2023-04-27 18:37:26 +01:00
Edward Hervey
88353d8cb2 qtdemux: Fix av1C parsing
This is a regression introduced by
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3882

The av1c codec configuration parsing would always fail due to an off-by-one
error, the content of an atom starting at offset 8 (i.e. the 9th byte) and not
9 (the 10th byte).

Also introduce a break in order to not get stray warnings

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4435>
2023-04-17 10:02:24 +01:00
Tim-Philipp Müller
1228ef095d multifile: error out if no filename was set
Fixes #2483

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4429>
2023-04-14 20:20:21 +00:00
Jan Alexander Steffens (heftig)
020115dc34 imagesequencesrc: Properly set default location
Noticed this because the generic_states test kept segfaulting at random.
GLibC 2.37 can crash when NULL is supplied as a format string.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4425>
2023-04-14 08:14:04 +00:00
Edward Hervey
75a550a1b1 twcc: Better handle duplicate packets
The previous code would only check if two packets in a row were duplicates. If
not (i.e. a packet is a duplicate of a packet received slightly before) the code
would generate completely bogus FCI because it assumes there were no duplicates
present in the array.

In order to be efficient, just store all received packets and remove the
duplicates just before the FCI is generated once the array of observations have
been sorted by seqnum.

Fixes TWCC usage with moderate to high packet duplication.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4378>
2023-04-10 13:16:44 +01:00
Sebastian Dröge
da4c5c01d1 rtspsrc: Skip PTs with caps incompatible to the global caps
Otherwise empty caps are created while all following code assumes that
the caps will have exactly one structure, and then run into assertions.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4343>
2023-04-05 00:56:05 +01:00
Tim-Philipp Müller
3c915cdca4 rtpjpegdepay: fix logic error when checking if an EOI is present
We wouldn't add the missing EOI marker if the frame ended with
either 0xFF NN or 0xNN D9.

Fixes #2407

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4269>
2023-03-25 01:00:36 +00:00
Piotr Brzeziński
6dad5345ea qtdemux: Fix seek adjustment with SNAP_AFTER flag
With GST_SEEK_FLAG_SNAP_AFTER present, the previous version would
adjust seek time based on the keyframe farthest away from desired_time.
This was incorrect, because we always want the *earliest* suitable keyframe
to seek to, not the last one.
With this fix, in case of the SNAP_AFTER, we now look for the closest keyframe
that can be found after desired_time. Behaviour for SNAP_BEFORE should remain
unchanged.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4251>
2023-03-22 16:51:16 +00:00
Sebastian Dröge
5d0d42fb95 matroskademux: Make gst_byte_reader_get_data() usage less confusing
This is effectively the same behaviour but retrieving 0 bytes of data is
confusing to read.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4221>
2023-03-18 19:52:40 +00:00
Arun Raghavan
5b83ba52b1 matroskamux: Set rate/channels in Opus template caps
For some reason these were missed, and if caps didn't have them, we would emit
an invalid Matroska file with a 0 value for Sampling Frequency or channels.

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2354
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4172>
2023-03-14 21:31:30 +00:00
Arun Raghavan
0dd60e06e8 rtpopusdepay: Assume 48 kHz if sprop-maxcapturerate is missing
This matches 7587, section 6.1:

>   sprop-maxcapturerate:  a hint about the maximum input sampling rate
>      [...]
>      bandwidths (Table 1).  By default, the sender is assumed to have
>      no limitations, i.e., 48000.

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2354
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4172>
2023-03-14 21:31:30 +00:00
Matt Feury
6b3adff951 rtspsrc: Consider "451: Parameter Not Understood" when handling broken control urls
similar to https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3854

it seems that some implementations return this when
the server does not implement URL handling correctly

this fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2334

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4129>
2023-03-07 18:15:25 +00:00
Sebastian Dröge
3ce43c8014 rtspsrc: Use the correct vfunc for the push-backchannel-sample action signal
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/446

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4056>
2023-02-23 12:54:28 +00:00
Seungha Yang
0e01e55778 qtmux: Fix assertion on caps update
GstQTMuxPad.configured_caps should be protected since it's
updated from streaming thread and accessed in aggregate thread

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4055>
2023-02-23 12:06:57 +00:00
Sebastian Dröge
1ccc256948 qtmux: Implement writing of av1C version 1 box
Version 0 is ancient and not specified in any documents. Take it
directly from the `codec_data` if presents or otherwise try to construct
a reasonably looking `av1C` box.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4027>
2023-02-21 17:45:34 +00:00
Sebastian Dröge
6d2bc8b8cd qtdemux: Drop av1C version 0 parsing and implement version 1 parsing
The av1C box is optional so dropping parsing does not break anything
fundamentally, and there seems to be no historical record how version 0
even looks like while the comments and the parsing disagreed with each
other.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4027>
2023-02-21 17:45:34 +00:00
Enrique Ocaña González
be4dc2d05f qtdemux: Don't emit GstSegment correcting start time when in MSE mode
When using qtdemux in a pipeline that should only work as a pure demuxer (not
for actual playback), qtdemux shouldn't emit new GstSegments to correct
the start time (jump to the future) to ensure that the user experiences no
playback delay. By doing so, it's generating the wrong segments when an append
of data from the past happens. When that happens, downstream elements such as
parsers (eg: aacparse) may clip those buffers laying before the GstSegment and
create problems on the GStreamer client app (eg: WebKit).

Getting buffers clipped out because of the wrong GstSegments started becoming
a problen when this commit was introduced:

ab6e49e9cc audioparsers: add back segment clipping to parsers that have lost it

This clipping makes test DASH shaka 35 from MVT tests[1] to fail in
WebKitGTK/WPE (at least) and can potentially cause a number of other problems
in the WebKit Media Source Extensions (MSE) code.

Note that this new behaviour of not emitting new GstSegments only makes sense
when qtdemux is being used as a pure demuxer and not as part of a regular
pipeline. This is why the variant field has been added. When equal to
VARIANT_MSE_BYTESTREAM, it will make qtdemux behave differently in push mode,
taking decisions that meet the expectations for an MSE-like processing mode.
This kind of tweaks have been done in the past for MSS streams, for instance.
That code has been refactored to use VARIANT_MSS_FRAGMENTED now, instead of
its own dedicated boolean flag.

Co-authored by: Alicia Boya García <ntrrgc@gmail.com>

...who suggested to use "variant: mse-bytestream" in the caps to identify that
mode, as proposed in her unmerged patch:

https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/467

[1] https://github.com/rdkcentral/mvt

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3990>
2023-02-18 10:38:30 +00:00
Vivia Nikolaidou
625f9aab09 qtdemux: Handle moov atom length=0 case by reading until the end
Previously it would fail to demux the file by trying to read G_MAXUINT64
bytes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3938>
2023-02-11 13:31:26 +00:00
Vivia Nikolaidou
cab020b4cb qtdemux: Fix guint vs gsize type confusion
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3938>
2023-02-11 13:31:26 +00:00
Sebastian Dröge
6ce76c43cb 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/3864>
2023-02-02 00:26:03 +00:00
Guillaume Desmottes
707156653f rtpptdemux: set different stream-id on each src pad
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3866>
2023-02-01 17:46:29 +00:00