Commit graph

763 commits

Author SHA1 Message Date
Edward Hervey
dd3542aa4d adaptivedemux2: Don't blindly set the main manifest URI as referer
There's no guarantee it will *actually* be the URI which refered to what we are
downloading. It could be a stream URI or anything else.

Instead of putting something wrong, put no (specific) referer as a better choice

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3972>
2023-03-20 07:59:27 +00:00
Edward Hervey
bead28ad5c hlsdemux2: Don't set a referer when updating playlists
In the same way we don't for regular playlists in the base class.

If there is a referer specified by the app/user, the downloadhelper will set it
accordingly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3972>
2023-03-20 07:59:26 +00:00
Sebastian Dröge
621ec7b6e8 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/4210>
2023-03-18 16:34:19 +02:00
Sebastian Dröge
7e2a0779c3 flacenc: Fix mapping of GStreamer image tag type to FLAC image tag type
These enums are not compatible so just casting them does not work.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4210>
2023-03-18 16:17:01 +02:00
Sebastian Dröge
ccad9a7338 plugins: Fix various trivial clang compiler warnings
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4210>
2023-03-18 16:16:55 +02:00
Enrique Ocaña González
735dac9d2f qtdemux: Fix crash on MSE-style flush
The flowcombiner and active_streams shouldn't be cleared in the
mse-bytestream variant, only in the mss-fragmented one. Otherwise the
soft reset leaves qtdemux in a state where it still believes that it has
streams, but they've been cleared. In that case, a null pointer
dereference happens and the app crashes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4199>
2023-03-17 15:33:49 +00:00
Tim-Philipp Müller
0fc568c6b1 gst-plugins-good: re-indent with GNU indent 2.2.12
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4182>
2023-03-17 03:18:54 +00:00
Arun Raghavan
82b892ba3e 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/4151>
2023-03-14 11:09:08 -04:00
Arun Raghavan
0ed51294e0 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/4151>
2023-03-14 11:09:08 -04:00
Itamar Marom
b8730bc98e splitmuxsink: Fix docs support version
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4138>
2023-03-09 15:08:19 +02:00
Matt Feury
224030ff0c 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/4123>
2023-03-07 10:32:32 -05:00
Seungha Yang
40300172ad adaptivedemux2: Fix MSVC build error
downloadrequest.c(497): error C4013: 'atoi' undefined; assuming extern returning int

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4107>
2023-03-03 23:15:42 +09:00
Alicia Boya García
c1f4bd5a3f qtdemux: Add MSE-style flush
The abort() method of SourceBuffer in Media Source Extensions is
expected to flush the demuxer and discard the current fragment,
if any. The configuration of tracks, if any, should be preserved.

qtdemux has different behavior for flush events depending on the
context.

This patch activates the intended behaviour only for streams of the
VARIANT_MSE_BYTESTREAM type, conformant to the ISO BMFF Bytestream
specification[1]. This flush behaviour is the same as the one
already in use for adaptivedemux sources.

[1] https://www.w3.org/TR/mse-byte-stream-format-isobmff/

https://bugzilla.gnome.org/show_bug.cgi?id=795424

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4101>
2023-03-02 17:54:41 +00:00
Shengqi Yu
83576690b6 matroskademux: Consider TrackUID==0 a warning and not handle it as error
some special files whose trackUID is 0 can be played on the other
player. But it cannot be played in GStreamer, because trackUID 0 will be
treated as an error in matroskademux.

So, it makes sense to only consider trackUID==0 a warning and not handle
it as error

https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1821

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4036>
2023-03-01 07:38:24 +00:00
Scott Kanowitz
2e4fd325e7 rtpsession: fix a race condition during the EOS event in gstrtpsession.c
This patch prevents a possible race condition from taking place between the EOS event handling and rtcp send
function/thread.

The condition starts by getting the GST_EVENT_EOS event on the send_rtp_sink pad, which causes two core things
to happen -- the event gets pushed down to the send_rtp_src pad and all sessions get marked "bye" prior to
completion of the event handler. In another thread the rtp_session_on_timeout function gets called after an
expiration of gst_clock_id_wait in the rtcp_thread function. This results in a call to the
ess->callbacks.send_rtcp(), which is configured as a function pointer to gst_rtp_session_send_rtcp via the
RTPSessionCallbacks structure passed to rtp_session_set_callbacks in the gst_rtp_session_init function.

In the race condition, the call to gst_rtp_session_send_rtcp can have the all_sources_bye boolean set to true
while GST_PAD_IS_EOS(rtpsession->send_rtp_sink) evaluates to false. This is the result of gst_rtp_session_send_rtcp
running before the send_rtp_sink's GST_EVENT_EOS handler completes. The exact point at which this condition occurs
is if there's a context switch to the rtcp_thread right after the call to rtp_session_mark_all_bye in the
GET_EVENT_EOS handler, but before the handler returns.

Normally, this would not be an issue because the rtcp_thread continues to run and indirectly call
gst_rtp_session_send_rtcp. However, the call to rtp_source_reset sets the sent_bye boolean to false, which ends up
causing rtp_session_are_all_sources_bye to return false. This gets passed to gst_rtp_session_send_rtcp and the EOS
event never gets sent.

The race condition results in the EOS event never getting passed to the rtcp_src pad, which prevents the bin and
pipeline from ever completing with EOS.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3798>
2023-02-28 17:01:08 +00:00
Sebastian Dröge
269915a51e 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/4050>
2023-02-23 09:22:23 +00:00
Seungha Yang
1f0528b428 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/4042>
2023-02-22 19:16:52 +00:00
Tim-Philipp Müller
517b0047e5 gst-plugins-good: update translations
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4040>
2023-02-22 12:22:12 +00:00
Rafał Dzięgiel
2d79f7d392 dashdemux2: mpdclient: Debug all restrictions when selecting rep
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3894>
2023-02-18 22:47:18 +01:00
Rafał Dzięgiel
d86b2d4efa dashdemux2: Add start-bitrate property
Similarly to hlsdemux2 that has this property, also add it to dashdemux2
so users can use it to choose first alternate.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3894>
2023-02-18 22:47:07 +01:00
Rafał Dzięgiel
9d720554a0 dashdemux2: Improve initial representation selection
Do not always start with lowest quality possible. Use properties set
by user to select best allowed initial representation at startup too.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3894>
2023-02-18 21:05:25 +00:00
Rafał Dzięgiel
38028c9873 hlsdemux2: Make start-bitrate property work without connection-speed
Makes "start-bitrate" work without setting "connection-speed" property. Having
another property set as a requirement for this one to work is unexpected.

This commit allows to request some initial bitrate for first segment, then
go into adaptive streaming for the rest of media playback.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3895>
2023-02-17 17:48:40 +01:00
Hosang Lee
0efb792fb4 tests: qtdemux: add test for MSS fragment wrong data offset compensation
A data offset with an offset smaller than the moof length is wrong
in smooth streaming streams. The samples will not be located and
eventually playback will error out. So compensate assuming data
is in mdat following moof.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3840>
2023-02-16 00:43:57 +00:00
Tim-Philipp Müller
491feead6e tests: qtdemux: use binary files for samples
Instead of hexdumping it in a 360k header file.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3840>
2023-02-16 00:43:57 +00:00
Hosang Lee
88f16ebd2a qtdemux: compensate wrong data offset for MSS fragments
A data offset with an offset smaller than the moof length is wrong
in smooth streaming streams.

The samples will not be located and eventually playback will
error out. So compensate assuming data is in mdat following moof.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3840>
2023-02-16 00:43:57 +00:00
Seungha Yang
f7c2602d41 splitmuxsrc: Proxy latency query to part reader
splitmuxsrc can respond to the latency query

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3566>
2023-02-15 23:47:50 +00:00
Khem Raj
817339c4de v4l2: Define ioctl_req_t for posix/linux case
this is an issue seen with musl based linux distros e.g. alpine [1]
musl is not going to change this since it breaks ABI/API interfaces
Newer compilers are stringent ( e.g. clang16 ) which can now detect
signature mismatches in function pointers too, existing code warned but
did not error with older clang

Fixes
gstv4l2object.c:544:23: error: incompatible function pointer types assigning to 'gint (*)(gint, ioctl_req_t, ...)' (aka 'int (*)(int, unsigned long, ...)') from 'int (int, int, ...)' [-Wincompatible-function-pointer-types]
    v4l2object->ioctl = ioctl;
                      ^ ~~~~~

[1] https://gitlab.alpinelinux.org/alpine/aports/-/issues/7580

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3950>
2023-02-14 20:36:28 +00:00
Vivia Nikolaidou
4e7a5ebb11 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/3934>
2023-02-11 02:20:39 +00:00
Vivia Nikolaidou
3a9acff978 qtdemux: Fix guint vs gsize type confusion
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3934>
2023-02-11 02:20:39 +00:00
Edward Hervey
f072b25940 adaptivedemux2: Use track ID for debugging
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3890>
2023-02-10 10:56:52 +00:00
Edward Hervey
5e193730db adaptivedemux2: Split track id from event stream-id
The id is used for naming of the various objects and debugging. We don't
want/need it to be obfuscated with the massive upstream id.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3890>
2023-02-10 10:56:52 +00:00
Sebastian Dröge
5486ed24a5 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/3882>
2023-02-09 14:04:06 +00:00
Sebastian Dröge
8593a58916 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/3882>
2023-02-09 14:04:06 +00:00
Patricia Muscalu
c3e52d5c4f rtph264pay: Don't insert SPS/PPS before the second image slice
Only the first slice, for which fist_mb_in_slice is set to 0,
should trigger insertion of SPS and PPS buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3402>
2023-02-08 12:10:11 +00:00
Enrique Ocaña González
92a4cfe20f 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/3867>
2023-02-06 12:42:49 +00:00
Edward Hervey
0639f117cb hlsdemux2: Remove enable-llhls property
This was only used for testing purposes

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:24 +00:00
Edward Hervey
854683c871 hlsdemux2: Don't leak PDT datetime
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:24 +00:00
Edward Hervey
96613c45fb adaptivedemux2: Don't leak taglist
Clarify the ownership in the documentation

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:24 +00:00
Edward Hervey
123030feac adaptivedemux2: Don't leak track tags
The tags are fully transfered to this function

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:23 +00:00
Jan Schmidt
6f6c0cbbaf adaptivedemux2: Log request duration in debug output
When completing, log how long a HTTP request took into the debug output.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:23 +00:00
Edward Hervey
714628f1ec hlsdemux2: Improve live playlist update intervals
The live playlists should be updated at a defined interval. The problem is that
this interval was used *after* the playlist was finally received and processed,
which resulted in a gradual shift happening in playlist updates.

Instead store and use the time at which playlists were requested to determine
when the next one should be downloaded.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:23 +00:00
Edward Hervey
6684aee14c hlsdemux2: Fix playlist reload interval when unchanged
When falling back to using the regular last segment, use that duration as the
identical-playlist reload interval (and not the playlist target duration which
could be much larger)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:23 +00:00
Edward Hervey
5935c8049a hlsdemux2: Fix position searching
The scanning is done in a reverse order, the proper full checks to do are
therefore:
* If the position is beyond half a "segment duration", it's in the following
segment
* If the position is within the first half of a segment, it's in that one
* If the segment is the first one and the position is within half a duration
backwards, we consider the position as being within that first segment

Also handle the case where a "partial only" segment doesn't have a reliable
duration, and therefore use the playlist target duration instead.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:23 +00:00
Edward Hervey
1c6364673d hlsdemux2: Handle all cases for starting segment calculation
The implementation wouldn't work with regular HLS streams (i.e. the final
fallback).

Now that the implementation uses time to search for the starting
segment (instead of just the n-th from the end), we can specify the correct
hold_back fallback value from the RFC

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:23 +00:00
Edward Hervey
3129970c8a hlsdemux2: Fix buffering threshold calculation and handling
* The checks for smaller values were wrong
* Properly initialize the stream default recommended buffering threshold so that
  a default (10s) value is used until the subclass can provide a proper value

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:23 +00:00
Edward Hervey
eb1eb64506 hlsdemux2: Make sure simple media playlist is properly primed
By setting/propagating stream time initially

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:23 +00:00
Jan Schmidt
3d0e8aa07e adaptivedemux2: Fix manifest access during seeking query
Avoid a deadlock if a downstream seeking query happens while the scheduler
thread is holding the manifest lock (for example during a seek back to live).

Instead, do a more elaborate fix where the external calls that need access to a
'manifest' access a copy that's updated during a manually triggered manifest
update callback.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:23 +00:00
Jan Schmidt
5334007a0b adaptivedemux2: Symbol hygiene cleanup
Rename track_dequeue_data_locked() to
gst_adaptive_demux_track_dequeue_data_locked(), since it's non-static.

Make find_stream_for_track_locked() static since it's only used in the main
gstadaptivedemux.c file.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:23 +00:00
Jan Schmidt
6bb74ed2a0 adaptivedemux2: Fix download error handling more
gst_adaptive_demux2_stream_finish_download() will already schedule another
fragment download if it can so don't fall through to the retry code that will
also try and schedule a download (triggering an assert).

Fix the logic in general to retry advancing into the live seek range once.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:23 +00:00
Jan Schmidt
b1354058e1 hlsdemux2: Immediately request playlist after URI changes
When the stream switches to a new playlist / variant while the loader is waiting
on a timer to refresh the old playlist, cancel the timer and submit the request
for the new URI.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:23 +00:00