There are cases when there is no demuxer involved that could do the
buffering, e.g. HLS with raw MP3 or AAC. In this case we want to place
the buffering multiqueue after the parser.
Before this change, we've considered the first element after the
adaptive streaming demuxer as a parser. This is not always true, e.g.
id3demux. Instead we now wait until we actually have a parser (or
decoder).
Fixes playback on such HLS streams.
Playbin3 takes lock when querying duration and handling
stream-collection message. So,to post stream-collection message,
duration query should be dropped when input pad is being unlinked.
https://bugzilla.gnome.org/show_bug.cgi?id=773341
max_buffer_usage is the index of the oldest buffer in the queue,
starting at zero, not the number of buffers queued.
find_limits returns the index of the oldest buffer that satisfies the
limits in its min_idx parameter, not the number of buffers needed. Fix
this use too in order to keep passing the tests that read
buffers-queued.
https://bugzilla.gnome.org/show_bug.cgi?id=775351
If a client gets dropped and the iteration gets restarted, bufpos is
incremented again for all clients that preceded the dropped one, causing
havoc.
Adjust the bufpos for all clients first before trying to drop any.
https://bugzilla.gnome.org/show_bug.cgi?id=774908
Optimize LE<->BE conversion by adding a dedicated fast path instead of
using the generic converter. Implement transform_ip function in order to do the
endian swap in place.
This saves buffer allocation for the intermediate format, can be done in place
and also performs the conversion in one step instead of unpack-convert-pack.
For all bit widths the naive algorithm is implemented, which provides the best
performance when compiled with -O3. ORC was considered but eventually removed
as it requires a dedicated function for in-place conversion (due to the
"restrict" parameters).
A more complex algorithm for the 24-bit conversion with unrolled loop and
32-bit processing is implemented in the #if 0 section. It performs better if
compiled with -O2. With -O3 however the naive algorithm performs better.
https://bugzilla.gnome.org/show_bug.cgi?id=773073
For frame->buffer, baseparse is doing that automatically for us. For
frame->output_buffer it doesn't and assumes that the subclass is already
doing that. Consistency!
Deterministic generation of snow and smpte is important for tests so
that it's not affected by other videotestsrc elements in current or
possibly previous tests.
https://bugzilla.gnome.org/show_bug.cgi?id=773102
find_suitable_mask() had complexity O(n^2) on the number of bits.
For common case like 2-channel audio the mask was calculated in about 4k loop
cycles.
Optimize both n_bits_set() and find_suitable_mask() to O(n) where n is the
number of bits set in the mask.
https://bugzilla.gnome.org/show_bug.cgi?id=772864
rawvideoparse wouldn't error out on not-negotiated,
but would just keep on going, because it didn't pass
the flow return value back to the parent class and
thus upstream, so the source wouldnt' stop streaming.
We have to calculate from the segment.stop, not the segment.start, as
playback goes from stop to start. This fix works around another race
condition in streamsynchronizer in my testcase.
See https://bugzilla.gnome.org/show_bug.cgi?id=771479
When connecting a demuxer through a multiqueue ensure to copy sticky
events in order to allow the following factory being properly
checked that it is functional.
https://bugzilla.gnome.org/show_bug.cgi?id=769580
https://github.com/mesonbuild/meson
With contributions from:
Tim-Philipp Müller <tim@centricular.com>
Jussi Pakkanen <jpakkane@gmail.com> (original port)
Highlights of the features provided are:
* Faster builds on Linux (~40-50% faster)
* The ability to build with MSVC on Windows
* Generate Visual Studio project files
* Generate XCode project files
* Much faster builds on Windows (on-par with Linux)
* Seriously fast configure and building on embedded
... and many more. For more details see:
http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.htmlhttp://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html
Building with Meson should work on both Linux and Windows, but may
need a few more tweaks on other operating systems.
This is enough for making it work in GES, but it's unclear if all the various
property combinations are working correctly. It's an improvement over what was
there before in any case, which was to just drop all buffers if rate < 0.0.
https://bugzilla.gnome.org/show_bug.cgi?id=769624
When processing EOS for a pad, send a stream-group-done
for the pad in case downstream is waiting for more
data on this stream before it can process related
streams from the group.
https://bugzilla.gnome.org/show_bug.cgi?id=768995
My collection leak fix 83f30627cd
introduced a crash in this scenario: audiotestsrc ! decodebin3 ! fakesink
The reference handling of collection in decodebin3 wasn't very clear and
my attempt to fix the leak introduced a regression where we went one
reference short in some other scenarios.
Fixing this by:
- Giving a strong reference to DecodebinInput making things clearer
- Fixing get_merged_collection() which was sometimes returning an
existing reference and sometimes a new one.
https://bugzilla.gnome.org/show_bug.cgi?id=769080
After we reset the resampler, there is no history anymore in the resampler
and the previously calculated output size is no longer valid.
Recalculate the new output size after a reset to make sure we don't try
to convert too much.
The collection owned by GstDecodebin3 has to be unreffed when disposing.
gst_event_new_stream_collection() doesn't consume the collection passed
to it so no need to give it an extra ref.
https://bugzilla.gnome.org/show_bug.cgi?id=768811
MultiQueueSlot owns a ref on the active stream so it should release it
when being freed.
DecodebinInputStream owns ref on the active and pending stream so they
should be dropped when being freed.
https://bugzilla.gnome.org/show_bug.cgi?id=768811
gst_stream_get_caps() returns a reffed caps.
The caps passed to gst_query_set_caps_result() are not transfered.
The caps in gst_parse_pad_stream_start_event() was either acquired
using gst_pad_get_current_caps() which returns a new ref or
explicitly reffed.
https://bugzilla.gnome.org/show_bug.cgi?id=768811
When a discont buffer is processed, the state is re-initialized, which
nullifies the allowed_tags.
The problem is when a subrip string with tags is processed and allowed_tags is
NULL. The function subrip_unescape_formatting() calls g_strjoinv with a
str_array as NULL, leading to a GLib-CRITICAL.
This patch removes the allowed_tags resetting, in parser_state_init(), but
move it into gst_sub_parse_format_autodetect().
https://bugzilla.gnome.org/show_bug.cgi?id=768525
With contributions from Jan Schmidt <jan@centricular.com>
* decodebin3 and playbin3 have the same purpose as the decodebin and
playbin elements, except make usage of more 1.x features and the new
GstStream API. This allows them to be more memory/cpu efficient.
* parsebin is a new element that demuxers/depayloads/parses an incoming
stream and exposes elementary streams. It is used by decodebin3.
It also automatically creates GstStream and GstStreamCollection for
elements that don't natively create them and sends the corresponding
events and messages
* Any application using playbin can use playbin3 by setting the env
variable USE_PLAYBIN3=1 without reconfiguration/recompilation.
We take a ref before removing which was never freeded.
The element is still alive anyway because the group has its own ref as
well.
Fix a leak with the 'test_suburi_error_wrongproto' test.
https://bugzilla.gnome.org/show_bug.cgi?id=766515
This helps in cases where raw audio data is being delivered, but the
buffers do not come in sample aligned sizes. The new unalignedaudioparse
bin can be autoplugged and configures an internal audioparse element to
align the data. audioparse itself gets support for audio/x-unaligned-raw
input caps; the output caps then contain the same information, except that
the name is changed to audio/x-raw (since audioparse aligns the data).
This ensures that souphttpsrc ! audioparse still works.
https://bugzilla.gnome.org/show_bug.cgi?id=689460
When we initialize an element in decodebin, we 1) set it to PAUSED and
push sticky events on its sinkpad to trigger negotiation 2) block its
src pad(s) to detect CAPS events. We can't block before 1) as that
would lead to a deadlock.
It's possible (and common) tho that an element configures its srcpad
during 1) and before 2). Therefore before this change we would
typically block and expose an element's pad only once the element
output its first buffer, triggering sticky events to be resent. One
consequence of this behaviour is that it sometimes broke
renegotiation.
With this change now we consider a pad ready to be exposed when it's
->blocked or has fixed caps (which were set before we could block it).
https://bugzilla.gnome.org/show_bug.cgi?id=765456
If we are configured to use buffering and there is no demuxer in the chain, we
still want a multiqueue, otherwise we will ignore the use-buffering property.
In that case, we will insert a multiqueue after the parser or decoder - not
elsewhere, otherwise we won't have timestamps.
https://bugzilla.gnome.org/show_bug.cgi?id=764948
gstsubparse.c: In function ‘parse_subrip’:
gstsubparse.c:988:7: error: ignoring return value of ‘strtol’, declared with attribute warn_unused_result [-Werror=unused-result]
cc1: all warnings being treated as errors
https://bugzilla.gnome.org/show_bug.cgi?id=765042
When blocking the subtitle pad, it's expected that stream-start
is the first event, and that it can precede caps arriving on the
peer pad - in fact the caps can only have arrived on the peer
pad when it was pre-primed with sticky events previously.
Instead, just pass the stream-start and don't block, because
stream-start is sticky anyway.
Don't require a cue identifier preceding the time range line
when parsing WebVTT. We could also store the CueID, but it's
not using anywhere, so just ignore it for now.
Make writable the buffer before pushing it lead to a buffer copy. It's
because a reference is keep for the previous buffer.
The previous buffer reference is only need to duplicate the buffer. In
drop-only mode, the previous buffer is release just after pushing the
buffer so a copy is done but it's useless.
https://bugzilla.gnome.org/show_bug.cgi?id=764319
Insert extra checks for the validity of the incoming
data when parsing subrip/webvtt content and debug log
output for invalid content.
Should fix Coverity warnings.
Remove some unused variables from the inner product functions.
Make filter coefficients by interpolating if required.
Rename some fields.
Try hard to not recalculate filters when just chaging the rate.
Add more proprties to audioresample.
Remove the consumed/produced output fields from the resampler and
converter. Let the caler specify the right number of input/output
samples so we can be more optimal.
Use just one function to update the converter configuration.
Simplify some things internally.
Make it possible to use writable input as temp space in audioconvert.
WebVTT is a new subtitle format for HTML5 video. In this first
version of the parser the cue settings are parsed but only stored in
the internal parser state structure. Later on these settings could be
part of the GstBuffer metadata.
https://bugzilla.gnome.org/show_bug.cgi?id=629764
There's a small window between decodebin choosing a buffering level
to post and another thread choosing a different buffering level
where things can race. Close that window by holding a new lock
that's only for posting buffering messages - like what was done
in multiqueue.
https://bugzilla.gnome.org/show_bug.cgi?id=764020
In check_upstream_seekable function, it returns FALSE value even though
we already declare about the seekable variable. So, This patch return
result of seekable in check_upstream_seekable function.
https://bugzilla.gnome.org/show_bug.cgi?id=763975
Due to transient locked state during autoplugging, some elements might be
ignored by the GstBin::change_state() and might still be running. Which could
then cause pad-added and similar accessing decodebin state that does not exist
anymore, and crash.
https://bugzilla.gnome.org/show_bug.cgi?id=763625
And also consider HEADER buffers without DELTA_UNIT flag as sync points. This
fixes sync-mode=2 with mpegtsmux for example, which has no streamheaders but
puts the HEADER flag on its keyframes.
https://bugzilla.gnome.org/show_bug.cgi?id=763278
In other places we lock it the other way around, leading to possible
deadlocks. Also this will deadlock if analyze_pad() causes a new element to be
autoplugged that adds new pads on itself when its state is changed.
https://bugzilla.gnome.org/show_bug.cgi?id=763491
This reverts commit 0615794300.
deinterlace was ported at some point in the last 4 years and has better video
format support, and especially better negotiation than avdeinterlace. Having
avdeinterlace but not deinterlace causes various problems in zerocopy
scenarios.
https://bugzilla.gnome.org/show_bug.cgi?id=760553
libgstreamer currently exports some debug category
symbols GST_CAT_*, but those are not declared in any
public headers.
Some plugins and libgstvideo just use GST_DEBUG_CATEGORY_EXTERN()
to declare and use those, but that's just not right at
all, and it won't work on Windows with MSVC. Instead look
up the categories via the API.
