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.
Avoids some false positives leading to miss identification:
* Prevent picture start code emulation for the first 2 bytes read
* Add check for valid "picture coding type" and "PB-frames mode" combination
Additionally, change name on confusingly named TR var to what
it is, the layer's PTYPE.
https://bugzilla.gnome.org/show_bug.cgi?id=693263
When getting caps of the decode chain, in get_topology, the caps are being
checked if fixed or not. But get_topology will be called when the decode is
chain is being exposed and hence it will always be fixed. Hence removing the
check for fixed caps. Removing gst_pad_get_current_caps for the chain->pad, as
get_pad_caps will again call the same api.
And get_topology can return NULL value if currently shutting down the
pipeline, which on being passed to create message will result in assertion
error. Check if topology is valid before using it
https://bugzilla.gnome.org/show_bug.cgi?id=755918
Avoid overflow in rate calculation. This can cause the resampler to
start on the wrong phase after a rate change.
Avoid overflow in cubic fraction calculation. This can cause noise when
dealing with higher samplerates.
analyze_new_pad() can return a new decode chain, which might have a new
GstDecodePad in the end. We should use those two for expose_pad() and not the
original ones that were passed to analyze_new_pad().
This fails when having a demuxer element that has raw pads immediately or
if a decoder with raw caps is after an adaptive demuxer.
https://bugzilla.gnome.org/show_bug.cgi?id=760949
[..] when resetting group start time. In GES, we are usually connected
to the streamsynchronizer on one audio and one video pad.
When seeking the timeline, both nlecompositions often output their flush_start
before any of them has output its flush_stop.
The current code, when receiving the first flush stop was using the
running time of the start of the second composition, which could
be pretty much anything, and means nothing at that point.
This patch is thread-safe, as STREAM_SYNCHRONIZER_LOCK is taken
both when setting flushing and when checking it.
https://bugzilla.gnome.org/show_bug.cgi?id=750013
When blocking input pads, we also need to properly set the appropriate
pending flag.
Without this, when switching stream types after initial configuration
(like going from Audio+Video to Audio+Video+Sub) playsink would never
wait for *all* input streams to be blocked (it would just wait for the
new input pad (text in this case) to be blocked).
Since the reconfiguration might introduce unlinking/relinking of elements,
we need to ensure that *ALL* input streams are blocked.
Failure to do so would result in having some input streams pushing data
to inactive elements (returning GST_FLOW_FLUSHING) or unlinked pads
(returning GST_FLOW_NOT_LINKED).
A later optimization could involve only blocking the input pads that
might be involved in reconfiguration. But better be safe than sorry for
now :)
Elements usually require that all fields on their caps are present
on the fixed caps they receive. Using intersection won't verify it,
resort to using is_subset() checks.
https://bugzilla.gnome.org/show_bug.cgi?id=760477
Those accept caps are actually checking if downstream supports
some particular caps to check if it need to negotiate a different
format. Checking only the next element with accept-caps is not enough
to guarantee that it is supported.
Using a caps query makes it obtain the supported caps for downstream
as a whole instead of only the next element.
Pass flags in _converter_new() so that we can configure ourselves
differently depending on some options.
SOURCE_WRITABLE -> IN_WRITABLE because the array is called 'in'
Simplify the API, we don't need the consumed and produced output
arguments. The caller needs to use the _get_in_frames/get_out_frames API
to check how much input is needed and how much output will be produced.
In this specific case it wouldn't cause problems as we only ever access the
first array element, but let's make explicit what is happening here.
CID 1346530 and 1346529
The filters' floating references are sinked during set_property() already,
which means that GstBin takes a new reference when adding the filter to it.
Get rid of the additional reference after adding the filter to the bin.
Unconditionally adding the template caps when proxying the caps query will play
havoc with decoders that attempt to choose an output format based on some caps
features. Creating a sink that does not include those caps features and a
decoder/parser/etc that preferentially chooses some specific caps feature when
available, will always return the decoder/parser/etc template caps and choose a
feature that downstream will be unable to support.
Fix by limiting the addition of the template caps to when the result is actually
empty.
https://bugzilla.gnome.org/show_bug.cgi?id=758212
This reverts commit 77dc09c3a9.
It can cause the FLUSH_START/STOP events to go to the sink elements, which
then causes state changes and various other problems. We shouldn't really
flush downstream here, the idea is to do *draining*.
Apart from that the testcase for the original bug here works without this
commit now.
Since the loops increasing count from 0 are always run at least
once (if count < 1), count will always be at least one when
compared to the drop/dup conditions.
Coverity 1139674
Add a property and logic to send a GstNetworkMessage event containing
the message that was received from a client. This can be used to
implement simply bidirectional communication.
Add a property and logic to send a GstNetworkMessageDispatched
event upstream to notify that a buffer has been sent. This can be used
to keep track of what client received what buffers.
Add a property to handle GstNetworkMessage events. These events contain
a buffer that is sent on the socket to allow for simple bidirectional
communication.
Otherwise we'll remove that element while keeping its buffering message in our
list, and because of that never ever report buffering 100% as that element
will always be at a lower percentage.
This fixes e.g. seeking over Period boundaries in DASH and various other
issues when buffering happens between group switches.
Also use a new mutex for protecting the buffering messages. The object lock is
already used by gst_object_has_as_ancestor() and we need to use it now for
checking if the buffering message sender has the to-be-removed element as
ancestor.
When we stop sending because we need more data, still keep a GSource
around to receive data from the clients.
Also handle read and write in the same go.
Make sure that any access to the GstDecodeBin's decode_chain
field is protected using the EXPOSE_LOCK. Also add a simple
reference counter to the GstDecodeChain structure so that when
the type_found signal fires it can hold onto the decode chain
even while the EXPOSE_LOCK is not held. This should fix a
race condition if the type_found signal fires right in the
middle of a state change that messes with the same decode
chain.
https://bugzilla.gnome.org/show_bug.cgi?id=755260
Just setting the ghostpad as flushing wasn't enough. It needs to be
consistent on the internal proxypad also, otherwise you end up in
situations where:
* a pending buffer on the target pad triggers the sticky event
propagation
* the default implementation sees that the proxypad is not flushing,
so it tries to push it to the other pad (the actual ghostpad)
* the ghostpad is flushing, so returns FALSE
* the push_event function sees that pushing the event failed...
* ... and pending buffer push returns GST_FLOW_ERROR, instead of
GST_FLOW_FLUSHING
By using gst_pad_set_active(FALSE), we ensure that both the ghostpad
and the proxypad are flushing/deactivated. The situation above will
no longer occur, and a GST_FLOW_FLUSHING will be returned.
Remove the format and layout from the mix_samples function and use the
format when creating the channel mixer object. Also use a flag to handle
the unlikely case of non-interleaved samples like we do elsewhere.
Add docs for the internal audioconvert object before moving it to the
audio library.
Remove get_sizes and implement the trivial logic in the element.
Remove some unused orc functions
Move the audio quantize code from audioconvert to the audio library.
work on making an audio converter helper function similar to the video
converter.
Fold fastrandom directly into the quantizer, add some ORC code to
optimize this later.
Rename _get_default_mask() to _get_fallback_mask() to make it more
clear that the function only provides a fallback if nothing else can be
done. Also clarify this in the documentation.
API: gst_audio_channel_get_fallback_mask()
In some conditions we might process empty buffers, calling
gst_control_binding_get_value_array in that case will lead
to the assertion:
(lt-ges-launch-1.0:18859): GStreamer-CRITICAL **: gst_control_binding_get_value_array: assertion 'values' failed
Add a TRUNCATE_RANGE flag for unpack functions to fill the least
significate bits with 0 (as did the old code). Also add functions
that don't truncate. Use the TRUNC flag in audioconvert for
backwards compatibility for now.
Use (1 << 31) as the multiplier for int<->float conversions. This makes
sure that int->float conversions always end up with floats between
[-1.0, 1.0].
For the conversion from float to int, this multiplier will give the complete
int range after we perform clipping.
Change the unit test to take this into consideration.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755301
The client-removed signal used G_INT_TYPE instead of G_SOCKET_TYPE
in its definition leading to problems on platforms where the size
of a pointer is larger than the size of an integer, It would also
not work at all with dynamic language bindings.
https://bugzilla.gnome.org/show_bug.cgi?id=757155
Rewrite audioconvert to try to make it more clear what steps are
executed during conversion.
Add passthrough step that just does a memcpy when possible.
Add ORC optimized dither and quantization functions.
Implement noise-shaping on S32 samples only and allow for arbitrary
noise shaping coefficients if we want this later.
