Simplify its usage by having it directly create the message if the collection
changed. This is what caller were always doing and avoids releasing selection
locks yet-another-time
Also use it in more places to avoid code repetition
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7002>
To simplify the description, I'm assuming we only have two streams: video and audio.
For the video stream, we have the following events :
- STREAM_START => stream->wait set to true
- NEW_SEGMENT(1) => blocked waiting in gst_stream_synchronizer_wait
- FLUSH_START => unblocked
- FLUSH_STOP => stream->wait reset to false
- NEW_SEGMENT(2) => not waiting, since stream->wait is false
Then for the audio stream, we have the following events :
- STREAM_START => stream->wait set to true
- NEW_SEGMENT(2) => blocked waiting in gst_stream_synchronizer_wait for ever.
Note: The first NEW_SEGMENT event and the FLUSH_START, FLUSH_STOP events of the audio stream
are dropped before being received by the streamsynchronizer element, because the decodebin audio pad src
is not yet linked to the playsink audio pad sink.
To fix this deadlock, we don't reset stream->wait to false in the FLUSH_STOP event when it is not
waiting for the EOS of the other streams.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6887>
The assertion that was present before is a bit too harsh, since there is now
a (understandable) use-case where this could happen.
In gapless use-case, with two files containing the same type (ex:audio). The
first one *does* expose a collection with an audio stream, but decoding
fails (for whatever reason).
That would cause us to have configured a audio combiner, which was never
used (i.e. not active).
Then the second file plays and we (wrongly) assume it should be activated
... whereas the combiner was indeed present.
Demote the assertion to a warning and properly handle it
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3389
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6742>
Since https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6153 ,
subtitle "decoders" (i.e. which decode to raw text) are no longer auto-plugged
by parsebin.
But if a given format does not have a parser at all, we would end up outputting
non-time/non-parsed outputs.
In order to mitigate the issue, until such parsers are available, we check if
the subtitle stream is in TIME format or not (i.e. whether it comes from a
parser or demuxer). If not, we attempt to plug in a subtitle "decoder".
Fixes#3463
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6597>
This will mimic the playbin2 behaviour, which sets the "next" entry to be
NULL.
The biggest impact this has is that when going back to READY the current play
entry will be discarded (instead of being kept around for when you go back to
PAUSED/PLAYING).
Fixes#3371
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6338>
find_slot_for_stream_id() will return a slot which has the request stream-id as
active_stream *or* pending_stream (i.e. the slot on which that stream is
currently being outputted or will be outputted).
When figuring out which slot to use (if any) we want to consider stream-id
which *will* appear on a given slot which isn't outputting anything yet the same
way as if we didn't find a slot yet.
Fixes races when doing intensive state changes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6332>
When dealing with demuxers which aren't streams-aware, we need to handle the
old-school "stream replacement" dance from `parsebin` and hide that in such a
way that output pads are re-used (if compatible).
By analyzing the collection posted by parsebin, we can:
* Identify whether some output slots are no longer used (because the stream they
currently handle is not present in the collection)
* Decide if some upcoming streams could re-use the existing slot
This supports both buffering and non-buffering modes.
Fixes#1651
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6201>
When switching urisourcebin, ensure that we first unlink *all* pads from
decodebin3 before linking them again.
This is to ensure that decodebin3 completely knows that all previous pads are no
longer needed and can prepare itself to being re-used.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6179>
Some subtitle "decoders" had a wrong category of "Parser", which `parsebin`
relies on to identify elements which do not *decode* streams but *parse* them.
This would cause such subtitle decoders to be plugged in within parsebin,
preventing the original stream to be properly used by (more efficient)
downstream decoders or subtitle renderers.
Fixes#1757
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6153>
It was adding and subtracting the segment base here and there, but it
was also doing so incorrectly, leading to various calculation errors.
Fixed a few bugs uncovered, related to getting a new segment:
* If we reset base_ts/next_ts/out_frame_count, also reset prevbuf
* Only do so if the new segment is different than the previous one
Also replaced a few occurrences of GST_BUFFER_TIMESTAMP with
GST_BUFFER_PTS for consistency.
Integrated the tests of
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2186
, now passing. The test_segment_update_same test had to be fixed,
because it was wrongly assuming that we would not fill the gap inside
the new-but-same segment.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6031>
In the following backtrace for the deadlock, we can see that:
- In T8 `uridecodebin3` is exposing a new pad, in `pad_added_cb`,
`playbin3` is trying to get `GST_PLAY_BIN3_LOCK` in the callback. This
threads holds its `SELECTION_LOCK` in F17 `reconfigure_output_stream`,
which is looks right `decodebin3` is handling its selection state
in that code path
- In T7 `playbin3` holds the `GST_PLAY_BIN3_LOCK` when calling
`gst_element_post_message` in `gst_play_bin3_send_event` which is
not necessary in that section of the code.
``` bt
Thread 8 (Thread 0x7f0b78ee36c0 (LWP 2952467) "multiqueue0:src"):
#0 futex_wait (private=0, expected=2, futex_word=0x1fa6d60) at ../sysdeps/nptl/futex-internal.h:146
#1 __GI___lll_lock_wait (futex=futex@entry=0x1fa6d60, private=0) at lowlevellock.c:49
#2 0x00007f0b858cd46a in lll_mutex_lock_optimized (mutex=0x1fa6d60) at pthread_mutex_lock.c:48
#3 ___pthread_mutex_lock (mutex=0x1fa6d60) at pthread_mutex_lock.c:128
#4 0x00007f0b7e665720 in pad_added_cb (uridecodebin=0x1fb4050, pad=0x7f0b54022060, playbin=0x1fb00e0) at ../subprojects/gst-plugins-base/gst/playback/gstplaybin3.c:2463
#5 0x00007f0b85c00060 in g_closure_invoke (closure=0x1fa9eb0, return_value=0x0, n_param_values=2, param_values=0x7f0b78ee1dd0, invocation_hint=0x7f0b78ee1d50) at ../gobject/gclosure.c:832
#6 0x00007f0b85c2cf66 in signal_emit_unlocked_R.isra.0 (node=node@entry=0x1c5bf30, detail=detail@entry=0, instance=instance@entry=0x1fb4050, emission_return=emission_return@entry=0x0, instance_and_params=instance_and_params@entry=0x7f0b78ee1dd0) at ../gobject/gsignal.c:3796
#7 0x00007f0b85c1d4da in g_signal_emit_valist (instance=<optimized out>, signal_id=<optimized out>, detail=<optimized out>, var_args=var_args@entry=0x7f0b78ee1f90) at ../gobject/gsignal.c:3549
#8 0x00007f0b85c1d6f3 in g_signal_emit (instance=<optimized out>, signal_id=<optimized out>, detail=<optimized out>) at ../gobject/gsignal.c:3606
#9 0x00007f0b85e20c3e in gst_element_add_pad (element=0x1fb4050, pad=0x7f0b54022060) at ../subprojects/gstreamer/gst/gstelement.c:802
#10 0x00007f0b7e632620 in add_output_pad (dec=0x1fb4050, target_pad=0x7f0b6400fda0) at ../subprojects/gst-plugins-base/gst/playback/gsturidecodebin3.c:717
#11 0x00007f0b7e632788 in db_pad_added_cb (element=0x1fb8020, pad=0x7f0b6400fda0, dec=0x1fb4050) at ../subprojects/gst-plugins-base/gst/playback/gsturidecodebin3.c:736
#12 0x00007f0b85c00060 in g_closure_invoke (closure=0x1fb7fc0, return_value=0x0, n_param_values=2, param_values=0x7f0b78ee2300, invocation_hint=0x7f0b78ee2280) at ../gobject/gclosure.c:832
#13 0x00007f0b85c2cf66 in signal_emit_unlocked_R.isra.0 (node=node@entry=0x1c5bf30, detail=detail@entry=0, instance=instance@entry=0x1fb8020, emission_return=emission_return@entry=0x0, instance_and_params=instance_and_params@entry=0x7f0b78ee2300) at ../gobject/gsignal.c:3796
#14 0x00007f0b85c1d4da in g_signal_emit_valist (instance=<optimized out>, signal_id=<optimized out>, detail=<optimized out>, var_args=var_args@entry=0x7f0b78ee24c0) at ../gobject/gsignal.c:3549
#15 0x00007f0b85c1d6f3 in g_signal_emit (instance=<optimized out>, signal_id=<optimized out>, detail=<optimized out>) at ../gobject/gsignal.c:3606
#16 0x00007f0b85e20c3e in gst_element_add_pad (element=0x1fb8020, pad=0x7f0b6400fda0) at ../subprojects/gstreamer/gst/gstelement.c:802
#17 0x00007f0b7e6260b4 in reconfigure_output_stream (output=0x7f0b5400def0, slot=0x7f0b64013dd0, msg=0x7f0b78ee26b8) at ../subprojects/gst-plugins-base/gst/playback/gstdecodebin3.c:3086
#18 0x00007f0b7e623700 in check_slot_reconfiguration (dbin=0x1fb8020, slot=0x7f0b64013dd0) at ../subprojects/gst-plugins-base/gst/playback/gstdecodebin3.c:2455
#19 0x00007f0b7e623e62 in multiqueue_src_probe (pad=0x7f0b6001e600, info=0x7f0b78ee2930, slot=0x7f0b64013dd0) at ../subprojects/gst-plugins-base/gst/playback/gstdecodebin3.c:2544
#20 0x00007f0b85e53aaa in probe_hook_marshal (hook=0x7f0b74040500, data=0x7f0b78ee28c0) at ../subprojects/gstreamer/gst/gstpad.c:3669
#21 0x00007f0b85c88a3e in g_hook_list_marshal (hook_list=0x7f0b6001e698, may_recurse=1, marshaller=0x7f0b85e53786 <probe_hook_marshal>, data=0x7f0b78ee28c0) at ../glib/ghook.c:674
#22 0x00007f0b85e541be in do_probe_callbacks (pad=0x7f0b6001e600, info=0x7f0b78ee2930, defaultval=GST_FLOW_OK) at ../subprojects/gstreamer/gst/gstpad.c:3853
#23 0x00007f0b85e5ac9c in gst_pad_push_event_unchecked (pad=0x7f0b6001e600, event=0x7f0b64002120, type=GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM) at ../subprojects/gstreamer/gst/gstpad.c:5538
#24 0x00007f0b85e54bc2 in push_sticky (pad=0x7f0b6001e600, ev=0x7f0b78ee2a60, user_data=0x7f0b78ee2ac0) at ../subprojects/gstreamer/gst/gstpad.c:4057
#25 0x00007f0b85e4a13c in events_foreach (pad=0x7f0b6001e600, func=0x7f0b85e54a8e <push_sticky>, user_data=0x7f0b78ee2ac0) at ../subprojects/gstreamer/gst/gstpad.c:613
#26 0x00007f0b85e54f91 in check_sticky (pad=0x7f0b6001e600, event=0x7f0b64002120) at ../subprojects/gstreamer/gst/gstpad.c:4116
#27 0x00007f0b85e5b65e in gst_pad_push_event (pad=0x7f0b6001e600, event=0x7f0b64002120) at ../subprojects/gstreamer/gst/gstpad.c:5706
#28 0x00007f0b7e5888e7 in gst_single_queue_push_one (mq=0x1fbb000, sq=0x7f0b64013b70, object=0x7f0b64002120, allow_drop=0x7f0b78ee2c3c) at ../subprojects/gstreamer/plugins/elements/gstmultiqueue.c:2068
#29 0x00007f0b7e58a1bc in gst_multi_queue_loop (pad=0x7f0b6001e600) at ../subprojects/gstreamer/plugins/elements/gstmultiqueue.c:2347
#30 0x00007f0b85e96a36 in gst_task_func (task=0x7f0b64016050) at ../subprojects/gstreamer/gst/gsttask.c:399
#31 0x00007f0b85e97e41 in default_func (tdata=0x7f0b640138d0, pool=0x1fbe9c0) at ../subprojects/gstreamer/gst/gsttaskpool.c:70
#32 0x00007f0b85cd1ab2 in g_thread_pool_thread_proxy (data=<optimized out>) at ../glib/gthreadpool.c:352
#33 0x00007f0b85ccc982 in g_thread_proxy (data=0x7f0b6006f640) at ../glib/gthread.c:831
#34 0x00007f0b858ca12d in start_thread (arg=<optimized out>) at pthread_create.c:442
#35 0x00007f0b8594bbc0 in clone3 () at ../sysdeps/unix/sysv/linux/x86_64/clone3.S:81
Thread 7 (Thread 0x7f0b7a7646c0 (LWP 2952434) "multiqueue3:src"):
#0 syscall () at ../sysdeps/unix/sysv/linux/x86_64/syscall.S:38
#1 0x00007f0b85cf470c in g_mutex_lock_slowpath (mutex=0x1fb81d0) at ../glib/gthread-posix.c:1494
#2 0x00007f0b7e6281a2 in gst_decodebin3_send_event (element=0x1fb8020, event=0x7f0b6800a650) at ../subprojects/gst-plugins-base/gst/playback/gstdecodebin3.c:3561
#3 0x00007f0b85e23ca3 in gst_element_send_event (element=0x1fb8020, event=0x7f0b6800a650) at ../subprojects/gstreamer/gst/gstelement.c:1994
#4 0x00007f0b7e63806b in gst_uri_decodebin3_send_event (element=0x1fb4050, event=0x7f0b6800a650) at ../subprojects/gst-plugins-base/gst/playback/gsturidecodebin3.c:2227
#5 0x00007f0b85e23ca3 in gst_element_send_event (element=0x1fb4050, event=0x7f0b6800a650) at ../subprojects/gstreamer/gst/gstelement.c:1994
#6 0x00007f0b7e66375a in gst_play_bin3_send_event (element=0x1fb00e0, event=0x7f0b6800a650) at ../subprojects/gst-plugins-base/gst/playback/gstplaybin3.c:1863
#7 0x00007f0b85e23ca3 in gst_element_send_event (element=0x1fb00e0, event=0x7f0b6800a650) at ../subprojects/gstreamer/gst/gstelement.c:1994
#8 0x00007f0b85f61b5b in stream_selection_cb (bus=0x1dc2d80, message=0x7f0b68008b00, d=0x1d7de30) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-scenario.c:2235
#9 0x00007f0b85c00060 in g_closure_invoke (closure=0x1d2a5b0, return_value=0x0, n_param_values=2, param_values=0x7f0b7a7627b0, invocation_hint=0x7f0b7a762730) at ../gobject/gclosure.c:832
#10 0x00007f0b85c2cf66 in signal_emit_unlocked_R.isra.0 (node=node@entry=0x1c5e5f0, detail=detail@entry=235, instance=instance@entry=0x1dc2d80, emission_return=emission_return@entry=0x0, instance_and_params=instance_and_params@entry=0x7f0b7a7627b0) at ../gobject/gsignal.c:3796
#11 0x00007f0b85c1d4da in g_signal_emit_valist (instance=<optimized out>, signal_id=<optimized out>, detail=<optimized out>, var_args=var_args@entry=0x7f0b7a762970) at ../gobject/gsignal.c:3549
#12 0x00007f0b85c1d6f3 in g_signal_emit (instance=<optimized out>, signal_id=<optimized out>, detail=<optimized out>) at ../gobject/gsignal.c:3606
#13 0x00007f0b85e05be9 in gst_bus_sync_signal_handler (bus=0x1dc2d80, message=0x7f0b68008b00, data=0x0) at ../subprojects/gstreamer/gst/gstbus.c:1307
#14 0x00007f0b85e03834 in gst_bus_post (bus=0x1dc2d80, message=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstbus.c:364
#15 0x00007f0b85e2436c in gst_element_post_message_default (element=0x1fb00e0, message=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstelement.c:2127
#16 0x00007f0b85df42b1 in gst_bin_post_message (element=0x1fb00e0, msg=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstbin.c:2789
#17 0x00007f0b85e24627 in gst_element_post_message (element=0x1fb00e0, message=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstelement.c:2170
#18 0x00007f0b85df7c12 in gst_bin_handle_message_func (bin=0x1fb00e0, message=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstbin.c:4041
#19 0x00007f0b85e61bd3 in gst_pipeline_handle_message (bin=0x1fb00e0, message=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstpipeline.c:669
#20 0x00007f0b7e663fa4 in gst_play_bin3_handle_message (bin=0x1fb00e0, msg=0x7f0b68008b00) at ../subprojects/gst-plugins-base/gst/playback/gstplaybin3.c:2030
#21 0x00007f0b85df5a98 in bin_bus_handler (bus=0x1dc2cc0, message=0x7f0b68008b00, bin=0x1fb00e0) at ../subprojects/gstreamer/gst/gstbin.c:3263
#22 0x00007f0b85e037f1 in gst_bus_post (bus=0x1dc2cc0, message=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstbus.c:357
#23 0x00007f0b85e2436c in gst_element_post_message_default (element=0x1fb4050, message=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstelement.c:2127
#24 0x00007f0b85df42b1 in gst_bin_post_message (element=0x1fb4050, msg=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstbin.c:2789
#25 0x00007f0b85e24627 in gst_element_post_message (element=0x1fb4050, message=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstelement.c:2170
#26 0x00007f0b85df7c12 in gst_bin_handle_message_func (bin=0x1fb4050, message=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstbin.c:4041
#27 0x00007f0b7e638005 in gst_uri_decode_bin3_handle_message (bin=0x1fb4050, msg=0x7f0b68008b00) at ../subprojects/gst-plugins-base/gst/playback/gsturidecodebin3.c:2218
#28 0x00007f0b85df5a98 in bin_bus_handler (bus=0x1dc2e40, message=0x7f0b68008b00, bin=0x1fb4050) at ../subprojects/gstreamer/gst/gstbin.c:3263
#29 0x00007f0b85e037f1 in gst_bus_post (bus=0x1dc2e40, message=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstbus.c:357
#30 0x00007f0b85e2436c in gst_element_post_message_default (element=0x1fb8020, message=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstelement.c:2127
#31 0x00007f0b85df42b1 in gst_bin_post_message (element=0x1fb8020, msg=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstbin.c:2789
#32 0x00007f0b85e24627 in gst_element_post_message (element=0x1fb8020, message=0x7f0b68008b00) at ../subprojects/gstreamer/gst/gstelement.c:2170
#33 0x00007f0b7e61ee43 in sink_event_function (sinkpad=0x7f0b6400f8c0, dbin=0x1fb8020, event=0x7f0b6c097870) at ../subprojects/gst-plugins-base/gst/playback/gstdecodebin3.c:1450
#34 0x00007f0b85f51122 in gst_validate_pad_monitor_downstream_event_check (pad_monitor=0x7f0b6c094a80, parent=0x1fb8020, event=0x7f0b6c097870, handler=0x7f0b7e61e797 <sink_event_function>) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-pad-monitor.c:2101
#35 0x00007f0b85f535bf in gst_validate_pad_monitor_sink_event_full_func (pad=0x7f0b6400f8c0, parent=0x1fb8020, event=0x7f0b6c097870) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-pad-monitor.c:2406
#36 0x00007f0b85f537fa in gst_validate_pad_monitor_sink_event_func (pad=0x7f0b6400f8c0, parent=0x1fb8020, event=0x7f0b6c097870) at ../subprojects/gst-devtools/validate/gst/validate/gst-validate-pad-monitor.c:2418
#37 0x00007f0b85e5c523 in gst_pad_send_event_unchecked (pad=0x7f0b6400f8c0, event=0x7f0b6c097870, type=GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM) at ../subprojects/gstreamer/gst/gstpad.c:5940
#38 0x00007f0b85e5ae65 in gst_pad_push_event_unchecked (pad=0x7f0b6400f650, event=0x7f0b6c097870, type=GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM) at ../subprojects/gstreamer/gst/gstpad.c:5573
#39 0x00007f0b85e54bc2 in push_sticky (pad=0x7f0b6400f650, ev=0x7f0b7a763620, user_data=0x7f0b7a763680) at ../subprojects/gstreamer/gst/gstpad.c:4057
#40 0x00007f0b85e4a13c in events_foreach (pad=0x7f0b6400f650, func=0x7f0b85e54a8e <push_sticky>, user_data=0x7f0b7a763680) at ../subprojects/gstreamer/gst/gstpad.c:613
#41 0x00007f0b85e54f91 in check_sticky (pad=0x7f0b6400f650, event=0x7f0b6c097870) at ../subprojects/gstreamer/gst/gstpad.c:4116
#42 0x00007f0b85e5b65e in gst_pad_push_event (pad=0x7f0b6400f650, event=0x7f0b6c097870) at ../subprojects/gstreamer/gst/gstpad.c:5706
#43 0x00007f0b85e523e7 in event_forward_func (pad=0x7f0b6400f650, data=0x7f0b7a763820) at ../subprojects/gstreamer/gst/gstpad.c:3130
#44 0x00007f0b85e521e3 in gst_pad_forward (pad=0x7f0b6004f180, forward=0x7f0b85e522bd <event_forward_func>, user_data=0x7f0b7a763820) at ../subprojects/gstreamer/gst/gstpad.c:3084
#45 0x00007f0b85e525ab in gst_pad_event_default (pad=0x7f0b6004f180, parent=0x7f0b6400f650, event=0x7f0b6c097870) at ../subprojects/gstreamer/gst/gstpad.c:3181
#46 0x00007f0b85e5c523 in gst_pad_send_event_unchecked (pad=0x7f0b6004f180, event=0x7f0b6c097870, type=GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM) at ../subprojects/gstreamer/gst/gstpad.c:5940
#47 0x00007f0b85e5ae65 in gst_pad_push_event_unchecked (pad=0x7f0b64008360, event=0x7f0b6c097870, type=GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM) at ../subprojects/gstreamer/gst/gstpad.c:5573
#48 0x00007f0b85e54bc2 in push_sticky (pad=0x7f0b64008360, ev=0x7f0b7a763a60, user_data=0x7f0b7a763ac0) at ../subprojects/gstreamer/gst/gstpad.c:4057
#49 0x00007f0b85e4a13c in events_foreach (pad=0x7f0b64008360, func=0x7f0b85e54a8e <push_sticky>, user_data=0x7f0b7a763ac0) at ../subprojects/gstreamer/gst/gstpad.c:613
#50 0x00007f0b85e54f91 in check_sticky (pad=0x7f0b64008360, event=0x7f0b6c097870) at ../subprojects/gstreamer/gst/gstpad.c:4116
#51 0x00007f0b85e5b65e in gst_pad_push_event (pad=0x7f0b64008360, event=0x7f0b6c097870) at ../subprojects/gstreamer/gst/gstpad.c:5706
#52 0x00007f0b7e5888e7 in gst_single_queue_push_one (mq=0x7f0b60076540, sq=0x7f0b6c093300, object=0x7f0b6c097870, allow_drop=0x7f0b7a763c3c) at ../subprojects/gstreamer/plugins/elements/gstmultiqueue.c:2068
#53 0x00007f0b7e58a1bc in gst_multi_queue_loop (pad=0x7f0b64008360) at ../subprojects/gstreamer/plugins/elements/gstmultiqueue.c:2347
#54 0x00007f0b85e96a36 in gst_task_func (task=0x7f0b6c072050) at ../subprojects/gstreamer/gst/gsttask.c:399
#55 0x00007f0b85e97e41 in default_func (tdata=0x7f0b6c093ef0, pool=0x1fbe9c0) at ../subprojects/gstreamer/gst/gsttaskpool.c:70
#56 0x00007f0b85cd1ab2 in g_thread_pool_thread_proxy (data=<optimized out>) at ../glib/gthreadpool.c:352
#57 0x00007f0b85ccc982 in g_thread_proxy (data=0x7f0b70033800) at ../glib/gthread.c:831
#58 0x00007f0b858ca12d in start_thread (arg=<optimized out>) at pthread_create.c:442
#59 0x00007f0b8594bbc0 in clone3 () at ../sysdeps/unix/sysv/linux/x86_64/clone3.S:81
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5982>
The internal elements are only created when caps on both video and subtitle pads
are known.
Prior to that, a GST_QUERY_CAPS on a video sink pad would just return ANY
instead of giving a hint of what downstream can actually handle and
prefers. This could result in upstream elements (such as decoders) deciding on
chosing (in the best cases) a non-optimal caps or (in the worst case) caps that
couldn't be handled by the elements downstream of subtitleoverlay.
In order to fix that, we assume that all subtitle "elements" handle the subtitle
overlay composition feature/meta and handle `GST_QUERY_CAPS` ourselves if the
internal elements aren't present yet.
Fixes#3176
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5834>
We access fields that are protected by the lock and this was already
held in other places where we call the method. I have got cases where
we get the following stack/assertion:
```
#0 g_logv (log_domain=0x7fb9d84e6cd5 "GStreamer", log_level=G_LOG_LEVEL_CRITICAL, format=<optimized out>, args=args@entry=0x7fb9d4de54e0) at ../glib/gmessages.c:1433
#1 0x00007fb9d802d0f3 in g_log (log_domain=<optimized out>, log_level=<optimized out>, format=<optimized out>) at ../glib/gmessages.c:1471
#2 0x00007fb9d845bc2c in gst_pad_send_event (pad=0x7fb98c01e050, event=0x7fb9c4105b90) at ../subprojects/gstreamer/gst/gstpad.c:6096
#3 0x00007fb9d6541c35 in gst_uri_decode_bin3_set_uri (dec=0x7fb9bc450960 [GstURIDecodeBin3], uri=0x7fb9c40f5410 "file:///var/home/thiblahute/devel/gstreamer/gstreamer/subprojects/gst-integration-testsuites/medias/defaults/mp4/mp3_h264.0.mp4") at ../subprojects/gst-plugins-base/gst/playback/gsturidecodebin3.c:1918
#4 0x00007fb9d6540c40 in gst_uri_decode_bin3_set_property (object=0x7fb9bc450960 [GstURIDecodeBin3], prop_id=1, value=0x7fb9d4de57b0, pspec=0x7fb9bcee5280 [GParamString]) at ../subprojects/gst-plugins-base/gst/playback/gsturidecodebin3.c:1569
#5 0x00007fb9d7f8f73d in object_set_property (object=0x7fb9bc450960 [GstURIDecodeBin3], pspec=0x7fb9bcee5280 [GParamString], value=0x7fb9d4de57b0, nqueue=0x7fb9c40d0c40, user_specified=<optimized out>) at ../gobject/gobject.c:1794
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5968>
In `parse_chain_output_probe()` the corresponding input stream might receive EOS
and thus be removed before the actual pad is removed. So we cannot assert about
this in `parsebin_pad_removed_cb()`.
Also, driving-by, protect `find_input_stream_for_pad()` with the selection lock
similarly to other functions accessing the input streams list.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5887>
This causes a lot of nasty side effects (like decoders assuming they are
actually linked downstream).
The reason why this was done was to check whether a decoder could handle the
actual caps, but this is the wrong way to do it.
The proper way to query whether a decoder can handle certain caps is via
`GST_QUERY_ACCEPT_CAPS` which is already done just before.
Partially reverts !4677 and partially fixes#3160
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5821>
Whenever that caps changes does not imply that a new segment will start.
Don't reset the last_ts if only the caps have changed. This fixes issues
if you have a stream without only first frame with TS=0, and have resolution
change happening. This was a regression introduced by !3059, which issue was
described in #1352. The reported issue is still fix after this change.
Fixes#1034
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5687>
The fake video decoder ignores input bitstream except
to enforce caps restrictions. It reads video width,
height and framerate from caps. Then it just pushes
video frames without doing any decoding.
The fake video decoder just draws a snake moving from
left to right in the middle of the frame. This is a
light weight drawing while it still provides an idea
about how smooth is the rendering.
The fake video decoder inherits from GstVideoDecoder.
It is useful to measure how smooth will be the whole
rendering pipeline if you had the most efficient video
decoder. Also useful to bisect issues for example when
suspecting issues in a specific video decoder.
Handles mpeg2, mpeg4, h263, h264, theora, vp8, wmv3, msmpeg,
flash-video, vp6, vp9, wmv1, wmv2, divx but more can be
added if needed.
For now it can only output RGBA, RGBx, BGRA, BGRx.
Its rank is 0 (none) but I added a property to change it so
that it can be selected by decodebin.
gst-launch-1.0 fakevideodec rank=512 \
playbin uri=http://clips.vorwaerts-gmbh.de/big_buck_bunny.mp4http://bugzilla.gnome.org/show_bug.cgi?id=723778Closes#679
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5636>
Update connection-speed at runtime in playbin, uridecodebin and decodebin
also do the same thing in urisourcebin.
With contributions from Philippe Normand <philn@igalia.com> (build fixes and
rebase on mono-repo).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4713>
Back in the mists of time[1], we switched `giostream*` elements to not close the
stream on stop() so that applications that needed a handle to the stream after
the element stopped had it.
Unfortunately, we also have cases[2] where waiting for the element to be
finalized is too late for the stream to be closed.
In order to not change the behaviour of the element, we add a property to allow
users to select the desired behaviour.
[1]: https://bugzilla.gnome.org/show_bug.cgi?id=587896
[2]: gst-plugins-rs#423
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5372>
The current limit is `x10`, which allows just `+20 dB` of gain.
While it may seem sufficient, this came up as a problem
in a real-world, non-specially-engineered situation,
in strawberry's EBU R 128 loudness normalization.
(https://github.com/strawberrymusicplayer/strawberry/pull/1216)
There is an audio track (that was not intentionally engineered that way),
that has integrated loudness of `-38 LUFS`,
and if we want to normalize it's loudness to e.g. `-16 LUFS`,
which is a very reasonable thing to do,
we need to apply gain of `+22 dB`,
which is larger than `+20 dB`, and we fail...
I think it should allow at least `+96 dB` of gain,
and therefore should be at `10^(96/20) ~= 63096`.
But, i don't see why we need to put any specific restriction
on that parameter in the first place, other than the fact
that the fixed-point multiplication scheme does not support volume
larger than 15x-ish.
So let's just implement a floating-point fall-back path
that does not involve fixed-point multiplication
and lift the restriction altogether?
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5063>
When switching from a raw stream to an encoded stream we need to make sure the
slot is unlinked, there is code in place for this but it wasn't triggered
because the slot being reconfigured wasn't advertised as linked beforehand.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5126>
Do not attempt to send a streams-selected message when reassigning
an output slot in case upstream signalled that it is handling stream selection.
In this case decodebin3 doesn't keep track of stream
collections (`dbin->collection` is NULL).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5059>
Setting the input field on the empty slot prevents future linking of it and will
result in flow errors later on.
This was observed in WebKit's MediaStream source element, when it changes the
caps on one of its associated streams, from an encoded format to a raw video
format. The associated stream-id on the sticky stream-start event doesn´t
change, but the element creates a new GstStream with a different ID and sets it
on the stream-start event. Stream parsing is disabled in urisourcebin, so
decodebin3 handles the parsing. Without this patch we would end-up with unlinked
pads in decodebin3 after switching to the raw video format.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5048>
This is a small optimization and avoids restarting the next parsing
iteration on already accepted data.
On its own it would also fix ZDI-CAN-20968 (see previous commit) but the
previous commit independently is also a valid fix for it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4895>
Add support for generation of 10/12/14/16 bit bayer test pattern.
The implementation is rather simplistic, just take the ARGB
input, generate 16-bit data out of it instead of 8-bit, shift
them as required by the output bitness, and apply endian swap.
Example usage:
```
$ gst-launch-1.0 videotestsrc ! \
video/x-bayer,width=512,height=512,format=bggr12le ! \
bayer2rgb ! \
video/x-raw,format=RGBA64_LE ! \
videoconvert ! \
autovideosink
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4686>
If the GST_MESSAGE_SRC of error message belongs to candidate decoders,
filter the error message and don't forward it as there might be a
following candidate decoder that can be used.
If the GST_MESSAGE_SRC of error message belongs to candidate decoders,
store the latency message and handle it after decoder is accepted.
This is to avoid the selection lock failure if decodebin3 needs to
handle latency message for candidate decoders when sending sticky event.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4677>
Send sticky events to the new created decoder after it switches
to PAUSED state. It it fails, just skip this decoder and try the
next one until finding one that works. Otherwise remove this
failing stream after trying all decoders and no one can work.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4677>
Generating the source element is done when urisourcebin is doing the READY to
PAUSED state change, so it is reasonable to set the new source element to that
state.
This also allows detecting early failures with backing libraries or
hardware (checks done in NULL->READY).
Finally it makes more sense to have an element in READY when attempting to query
information from it (such as SCHEDULING queries or probing live-ness).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3856>
self->eos was never reset after streamsynchronizer has sent EOS
(except on explicit flush or switching back to PAUSED).
As a result, synchronization was broken if new streams were pushed later
as gst_stream_synchronizer_wait() does not wait if self->eos is set.
Fix this by reseting self->eos on STREAM_START as that means a new
stream is being sent upstream and so a new EOS will follow later on.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4749>
While decodebin3 could handle changes in inputs (ex: changing codecs), there was
still one limitation which was when changing between sources which had
non-intersecting stream types (ex: switching from a video-only source to a
audio-only source). While the decoder *could* change to the proper codec ... it
would carry on using a `DecodebinOutputStream` associated to that stream
type (and therefore with pads with the wrong name).
In order to handle this:
* We notify the `MultiQueueSlot` of the change in `GstStreamType` if it already
had an associated inputstream (ex: the one associated with the static sink
pad)
* We detect such changes on the output of multiqueue as soon as
possible (i.e. when we get the GST_EVENT_STREAM_START for the new stream type)
by discarding the associated output.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1669
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4703>
