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4304 commits

Author SHA1 Message Date
Thibault Saunier
135c612550 encodebin: Allow streamheader update when profile.allow_dynamic_output == FALSE
Some encoders can update the stream header through time (for example
vp8 might do that) but it does not strictly changes the output format.
2016-01-27 12:58:23 +01:00
Sebastian Dröge
acd08a828d decodebin: Correctly expose pads from elements that have directly exposable pads
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
2016-01-25 13:50:26 +01:00
Mathieu Duponchelle
2717f4a86f streamsynchronizer: Ignore flushing streams [..]
[..] 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
2016-01-16 11:05:13 +01:00
Sebastian Dröge
fccf83e69f playbin: Only append non-raw and sysmem pad template caps to the autoplug-query result
Otherwise a decoder supporting GL memory will think that all downstream can
support GL memory because of seeing its own template caps.

https://bugzilla.gnome.org/show_bug.cgi?id=758212
2016-01-16 11:05:13 +01:00
Sebastian Dröge
9713ab06cd Revert "playbin: only add the template caps when the result is empty"
This reverts commit 023af2d3b1.

https://bugzilla.gnome.org/show_bug.cgi?id=758212
2016-01-16 11:05:13 +01:00
Edward Hervey
62053852de playsink: Properly mark pending blocked pads
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 :)
2016-01-15 10:05:58 +01:00
Thiago Santos
0d18717912 subtitleoverlay: replace gst_caps_can_intersect() with is_subset()
Subset check verifies also that all required fields are present
and is mostly commonly used when checking if an element accepts
a certain caps
2016-01-13 16:32:25 -03:00
Thiago Santos
81c52aaa16 playbin: use subset check instead of intersect
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
2016-01-13 15:29:17 -03:00
Thiago Santos
20f6af651b subtitleoverlay: replace accept-caps with caps query
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.
2016-01-11 18:35:29 -03:00
Thiago Santos
5ef0a09794 videorate: replace accept-caps with a caps query
accept-caps is only a shallow check, it needs to know
whether downstream as a whole accepts the framerate
2016-01-08 15:05:38 -03:00
Wim Taymans
85afad72ec audio-converter: small API tweaks
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'
2016-01-08 17:34:50 +01:00
Wim Taymans
980163457e audio-convert: simplify API
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.
2016-01-08 17:19:58 +01:00
Sebastian Dröge
844aa3e6a9 playbin: Use the caps query instead of accept-caps to detect if a sink accepts caps
accept-caps is only for one element, caps query is recursive. Fixes playback
with totem and other situations.

https://bugzilla.gnome.org/show_bug.cgi?id=760234
2016-01-08 16:32:32 +02:00
Aurélien Zanelli
9b9f913809 videotestsrc: add missing break in set_property switch case
To avoid future issue when adding new properties.

https://bugzilla.gnome.org/show_bug.cgi?id=760204
2016-01-06 13:21:06 +02:00
Sebastian Dröge
eb09889176 audioconvert: Pass pointer arrays instead of singleton pointers to gst_audio_converter_samples()
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
2015-12-29 18:14:54 +02:00
Sebastian Dröge
0416f121f2 typefindfunctions: Make sure that enough data is available in AAC/ADTS typefinder
We would otherwise read beyond the array bounds and crash every now and then.
This was introduced with 5640ba17c8.

https://bugzilla.gnome.org/show_bug.cgi?id=759910
2015-12-28 13:51:02 +02:00
Sebastian Dröge
6a57399270 playsink: Don't leak audio/video filters due to floating references weirdness
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.
2015-12-25 11:34:10 +01:00
Sebastian Dröge
a136ac0e2f playsink: Allow reuse of audio/video filters by unparenting them from their bins
And also recreate the chains if the filter is changing.
2015-12-25 10:36:44 +01:00
Sebastian Dröge
24181db083 playsink: Don't leak audio/video filters when using non-raw media 2015-12-25 10:28:02 +01:00
Matthew Waters
023af2d3b1 playbin: only add the template caps when the result is empty
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
2015-12-18 21:55:00 +11:00
Sebastian Dröge
60bad4815d Revert "decodebin2: fix deadlock on chain shutdown"
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.
2015-12-16 17:09:25 +01:00
Luis de Bethencourt
29cfb9a6d7 multifdsink: fix typo in GST_WARNING_OBJECT
This should make easier to parse the debug logs.
s/fnctl/fcntl
2015-12-16 11:12:03 +00:00
Vincent Penquerc'h
033ce9b20d videorate: remove dead code
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
2015-12-16 11:00:22 +00:00
Wim Taymans
8bcf183c7f audioconvert: clear convert object 2015-12-16 11:13:15 +01:00
Wim Taymans
f5a3f70571 audio: adapt API for non-interleaved formats
Allow an array of sample blocks to be passed to the channel mix and
quantizer functions to support non-interleaved formats.
2015-12-14 09:16:08 +01:00
Wim Taymans
9c2bcd7b76 multisocketsink: add GstNetworkMessage event
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.
2015-12-10 12:44:42 +01:00
Wim Taymans
9aaaa26ff3 multisocketsink: add dispatched event
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.
2015-12-10 12:44:42 +01:00
Wim Taymans
0e1a858d89 socketsrc: handle GstNetworkMessage events
Add a property to handle GstNetworkMessage events. These events contain
a buffer that is sent on the socket to allow for simple bidirectional
communication.
2015-12-10 12:44:42 +01:00
Wim Taymans
5e55968546 audio-convert: improve converter API
Improve the converter API to allow for an max input and output number of
samples and return the number of consumed/produced samples.
2015-12-09 17:16:26 +01:00
Wim Taymans
1da5a3ab66 multisocketsink: let downstream know we support metadata
Let downstream know that we support GstNetControlMessage metadata API.
2015-12-04 12:25:11 +01:00
Tim-Philipp Müller
71505dfa24 decodebin2: fix "Attempt to unlock mutex that was not locked"
Introduced in commit ee44337f, caused the decodebin
test_text_plain_streams unit test to abort.

https://bugzilla.gnome.org/show_bug.cgi?id=752651
2015-12-02 18:16:05 +00:00
Edward Hervey
d292ed48c5 playback: Expose XSUB formats by default
This is a workaround, we should remove this once we have a proper
decoder
2015-12-02 16:37:50 +01:00
Edward Hervey
c79bf13bc2 streamsynchronizer: Rename GstStream => GstSyncStream
Avoid clashes with future GstStream from core
2015-12-02 16:37:41 +01:00
Sebastian Dröge
9e4bf58b8e decodebin: Update buffering messages when removing an element that had buffering pending
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.
2015-12-02 16:16:22 +02:00
Wim Taymans
01f5ca3da8 multisocketsink: keep on reading when we stop sending
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.
2015-12-02 10:26:03 +01:00
Thomas Bluemel
2c62aad159 [PATCH] Fix a race condition accessing the decode_chain field.
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
2015-12-01 17:36:31 +00:00
Vincent Penquerc'h
870c6df489 decodebin: early out on pad-added when the pad is inactive
The pad may be recently deactivated if the element is switched
back down very quickly.

https://bugzilla.gnome.org/show_bug.cgi?id=752651
2015-12-01 17:36:31 +00:00
Vincent Penquerc'h
ee44337fc3 decodebin: lock the expose lock around decode_chain use
Helps with a crash in decodebin when quickly switching states.

https://bugzilla.gnome.org/show_bug.cgi?id=752651
2015-12-01 17:36:31 +00:00
Wim Taymans
ff6d1a2a25 audio-converter: add output size argument
Make it possible to have a different number of output samples than input
samples when we, for example, want to add resampling later.
2015-11-10 09:53:59 +01:00
Edward Hervey
d0eface01c decodebin: Properly deactivate ghostpads
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.
2015-11-06 19:38:13 +01:00
Tim-Philipp Müller
d2e210bbea audioconvert: fix build
Don't include file that is no longer generated, and remove some
files that are no longer needed because they have moved into the
lib. Fixes distcheck.
2015-11-06 18:12:28 +00:00
Wim Taymans
e3f0f3b91e audio-converter: move audio converter to audio libs
Move the audio-converter helper to the audio library.
2015-11-06 17:53:22 +01:00
Wim Taymans
dfa25a40fc audio-channel-mix: move channel mixer to audio libs
Move the channel mixer code to the audio library
2015-11-06 17:39:33 +01:00
Wim Taymans
b8bea9d8be audio: add debug categories 2015-11-06 17:29:22 +01:00
Wim Taymans
268ed5dd6f channelmix: don't limit channelpositions
Don't set a limit on the channel positions, just like the metadata.
2015-11-06 16:42:35 +01:00
Wim Taymans
9fbe0386d0 channelmix: simplify API a little
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.
2015-11-06 16:03:20 +01:00
Wim Taymans
7f5104f52f channelmix: GstChannel -> GstAudioChannel
Rename GstChannel to GstAudioChannel
2015-11-06 15:50:34 +01:00
Wim Taymans
1635bc0a45 audioconvert: cleanups and add some docs
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
2015-11-06 12:46:36 +01:00
Wim Taymans
c36ac3ce45 audioconvert: move audio quantize code to libs
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.
2015-11-06 12:10:48 +01:00
Wim Taymans
a7789854d5 audio-channels: rename get_default_mask
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()
2015-11-05 12:50:18 +01:00
Thibault Saunier
9c7d3c8ab2 volume: Do not try to get binding value array if we are not processing any sample
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
2015-11-05 11:44:31 +01:00
Wim Taymans
f86ed8cdf6 audio-channels: make method to get default channel-mask
Add a new method to get the default channel-mask.
Use the new method on audiodecoder and audioconvert.

API: gst_audio_channel_get_default_mask()
2015-11-05 10:52:53 +01:00
Wim Taymans
801f7ca464 audio-format: add TRUNCATE_RANGE flag
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.
2015-11-03 12:12:08 +01:00
Wim Taymans
9e15c89564 audioconvert: change multiplier for int<->float conversion
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
2015-11-03 12:12:08 +01:00
Wim Taymans
bd89f2430b audiotestsrc: increase freq limit
Raise the frequency limit and try to negotiate to a samplerate of 4*freq
when larger then the default samplerate.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=754450
2015-11-02 15:54:19 +01:00
Wim Taymans
c688eb0d88 audiotestsrc: add support for unlimited number of channels
Raise the channel limit and set the channel-mask for > 2 channels.
2015-11-02 15:46:22 +01:00
Wim Taymans
b0bf294a62 audiotestsrc: add support for all formats
Use the pack functions to also support the other audio formats we
have.
2015-11-02 13:22:18 +01:00
Sebastian Dröge
e51c9a3dad audioresample: Clip input buffers to the segment before handling them
https://bugzilla.gnome.org/show_bug.cgi?id=757068
2015-11-02 10:20:37 +02:00
Sebastian Dröge
000c424835 audioconvert: Clip input buffers to the segment before handling them
https://bugzilla.gnome.org/show_bug.cgi?id=757068
2015-11-02 10:20:37 +02:00
Tim-Philipp Müller
3dd26bb9e8 audioconvert: update orc backup code to fix build without orc 2015-11-01 23:06:11 +00:00
Csaba Toth
3159501002 multisocketsink: fix "client-removed" signal on 64-bit platforms and with bindings
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
2015-10-31 11:12:38 +00:00
Joan Pau Beltran
a95a900c21 videotestsrc: fix handling of Bayer format 'gbrg'
Due to a typo, videotestsrc did not handle the Bayer
format 'gbrg' properly and reported it as invalid,
causing negotiation errors.

https://bugzilla.gnome.org/show_bug.cgi?id=757264
2015-10-30 20:29:04 +00:00
Wim Taymans
5cf367ae57 audioconvert: rework audioconvert
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.
2015-10-30 17:51:47 +01:00
Wim Taymans
e1569ce76a channelmix: fix up API a little
don't use gpointer * for something that should be gpointer.
2015-10-30 17:51:47 +01:00
Wim Taymans
26d469a04b audioquantize: make helper for add with saturation 2015-10-30 17:51:47 +01:00
Wim Taymans
cd6c29e071 audioconvert: make the quantizer a reusable object
Turn the quantizer into a reusable object.
2015-10-28 11:36:18 +01:00
Wim Taymans
8fc2569328 audioconvert: make the channel mixer a separate reusable object
A first attempt at making the channel mixer a separate object.
2015-10-28 11:36:18 +01:00
Wim Taymans
8d4cd51e59 audioquantize: fix 8-pole noise shaping
Fix the 8-pole noise shaping error update. We were mixing errors from
different channels.
2015-10-28 11:36:18 +01:00
Sebastian Dröge
36b80edb72 decodebin: Send SEEK events directly to adaptive streaming demuxers
This makes sure that they will always get SEEK events, even if we're currently
in the middle of a group switch (i.e. switching to another
representation/bitrate/etc).

https://bugzilla.gnome.org/show_bug.cgi?id=606382
2015-10-27 15:50:45 +02:00
Guillaume Desmottes
7d6b6b0313 decodebin: fix event leak
As stated in GST_PAD_PROBE_HANDLED's documentation, we are
supposed to unref the event before returning.

Fixes an event leak in the validate.hls.playback.play_15s.hls_bibbop
validate scenario.

https://bugzilla.gnome.org/show_bug.cgi?id=754459
2015-10-25 11:18:29 +00:00
Sebastian Dröge
b4afaee8c0 audioconvert: Update disted orc files 2015-10-23 19:13:05 +03:00
Wim Taymans
2b626a5adf audioconvert: use pack/unpack functions
Rework the converter to use the pack/unpack functions
Because the unpack functions can only unpack to 1 format, add a separate
conversion step for doubles when the unpack function produces int.
Do conversion to S32 in the quantize function directly.
Tweak the conversion factor for doing float->int conversion slightly to
get the full range of negative samples, use clamp to make sure we don't
exceed our int range on the positive axis (see also #755301)
2015-10-23 16:58:17 +02:00
Sebastian Dröge
53f135cec7 playbin: Send upstream events directly to playsink
Send event directly to playsink instead of letting GstBin iterate
over all sink elements. The latter might send the event multiple times
in case the SEEK causes a reconfiguration of the pipeline, as can easily
happen with adaptive streaming demuxers.

What would then happen is that the iterator would be reset, we send the
event again, and on the second time it will fail in the majority of cases
because the pipeline is still being reconfigured
2015-10-23 12:02:28 +03:00
Thibault Saunier
ab6b536a66 videotestsrc: Force alpha downstream if foreground color contains alpha
Otherwise the foreground color won't be fully represented in the
outputted frames.

https://bugzilla.gnome.org/show_bug.cgi?id=755482
2015-10-22 11:12:23 +02:00
Matthew Waters
44871680f0 decodebin: track the exposable pads through connect_pad
The logic introduced by
[d50b713: decodebin: set the decode pad target before setting elements to PAUSED]
to expose pads would only ever be able to possibly expose one (the last) pad per element.

Make it so that any exposable pads are able to be exposed rather than just the
last pad returned by connect_element.

https://bugzilla.gnome.org/show_bug.cgi?id=742924
2015-10-20 10:48:05 +03:00
Matthew Waters
94d81fc713 decodebin: return the possibly new chain in analyze_new_pad
In the case of analyzing a demuxer chain, analyze_new_pad may create
a new GstDecodeChain.  This was not propagated to the calling function which as
of [d50b713f decodebin: set the decode pad target before setting elements to PAUSED]
is now required to be able to expose the correct pad.

https://bugzilla.gnome.org/show_bug.cgi?id=742924
2015-10-20 10:47:45 +03:00
Rajat Verma
68ec631db7 playsink: relink text_pad in case of reconfiguration
In case of reconfiguration, text_pad should be re-connected with
stream synchronizer sink pad. Otherwise we'll leave an unlinked pad around if
there always was a streamsynchronizer text pad.

https://bugzilla.gnome.org/show_bug.cgi?id=756804
2015-10-20 10:37:04 +03:00
Sebastian Dröge
4d6aa0f831 decodebin/playbin/playsink/subtitleoverlay: Post async-done on state change failures
https://bugzilla.gnome.org/show_bug.cgi?id=756611
2015-10-19 11:06:25 +03:00
Sebastian Dröge
87dbe54797 playsink: Immediately error out if state change fails
Otherwise we chain up to the parent class' change_state function and might
override the failure with SUCCESS.

https://bugzilla.gnome.org/show_bug.cgi?id=756611
2015-10-19 11:06:25 +03:00
Sebastian Dröge
92061cb19e playbin/uridecodebin: Always post async-done immediately if we're a live pipeline
Not only if the base class told us, but also if one of our own elements did.

https://bugzilla.gnome.org/show_bug.cgi?id=756611
2015-10-19 11:06:25 +03:00
Matthew Waters
d50b713f44 decodebin: set the decode pad target before setting elements to PAUSED
Otherwise caps and context queries will disappear into nothing and therefore
fail.  With autoplug-query now actually working, users (such as playbin) can
proxy these queries to the selected video sink and be able to select an
more appropriate configuration.

https://bugzilla.gnome.org/show_bug.cgi?id=731204
2015-10-19 11:55:04 +11:00
Vineeth TM
b424bc2e4f playsink: Fix volume element leak
In case sink implements a streamvolume interface, volume element is being got
from the sink. But this is transfer full. So the memory should be freed before
setting it to NULL. This was resulting in major memory leaks

https://bugzilla.gnome.org/show_bug.cgi?id=755867
2015-10-15 09:42:21 +03:00
Vineeth T M
a1d84edd16 videorate: remove unnecessary break statement
Trivial patch to remove unncessary break statement used after
goto statement.

https://bugzilla.gnome.org/show_bug.cgi?id=754054
2015-10-02 17:27:13 +03:00
Mathieu Duponchelle
0c0f803488 encodebin: Fix special case
Allows to run such a command line :

gst-launch-1.0 uridecodebin uri=file:///home/meh/Music/sthg.mp4 ! \
encodebin profile-string="audio/x-wav|1" ! filesink location=sthg.wav

Previously the code failed because wavenc is considered as a muxer.
We still want encodebin to audio/x-wav as an AudioEncodingProfile,
so this simple fix allows that.

Ability to mux raw streams in containers such as matroskamux
is a different issue.

https://bugzilla.gnome.org/show_bug.cgi?id=751470
2015-10-02 17:25:48 +03:00
Rajat Verma
267f4c2bad decodebin: free hidden groups at time of switching groups
hidden groups should be freed at time of switching groups to avoid memory use
from balloning up.

https://bugzilla.gnome.org/show_bug.cgi?id=755770
2015-10-02 17:02:21 +03:00
Jan Schmidt
3f7138a6e4 videotestsrc: Don't fixate framerate if downstream didn't provide one
intersection with a downstream that accepts any video/x-raw caps
with no further detail won't create a framerate field. If it's
not in the caps, don't fixate it, just set it to 30/1
2015-10-02 15:05:26 +10:00
Tim-Philipp Müller
fb30c04145 typefinding: minor clean-up
Remove unnecessary brackets from IS_MPEGTS_HEADER macro.
2015-10-01 12:49:59 +01:00
Pankaj Darak
eaf4ce01d3 typefinding: mpeg-ts detection improvement
Allow AFC to be 0 for null pid packets.

https://bugzilla.gnome.org/show_bug.cgi?id=726117
2015-10-01 12:32:33 +01:00
Tim-Philipp Müller
7fb9cd453b subparse: detect closing tags even if there's a space after the slash
</ i> should be handled like </i>

https://bugzilla.gnome.org/show_bug.cgi?id=755875
2015-09-30 18:17:13 +01:00
Vineeth T M
060f0c21f2 audiosink, multisocketsink: Fix error leak during failures
https://bugzilla.gnome.org/show_bug.cgi?id=755143
2015-09-17 11:59:35 +02:00
Sebastian Dröge
2a1e046dd9 uridecodebin: Use the correct caps name for MS Smooth Streaming manifests
Thanks to John Chang <r97922153@gmail.com> for reporting.

https://bugzilla.gnome.org/show_bug.cgi?id=755098
2015-09-16 19:54:43 +02:00
Sebastian Dröge
35cb3b0c57 playback: Add POINTER_TO_ULONG() macro for consistency 2015-09-11 23:29:57 +02:00
Kouhei Sutou
ab64b00b48 playback: fix build error for 64bit Windows build by MinGW
Casting to gpointer from gulong generates the following warning with
64bit Windows target MinGW:

    gstplaybin2.c: In function 'pad_added_cb':
    gstplaybin2.c:3476:7: error: cast to pointer from integer of different size [-Werror=int-to-pointer-cast]
           (gpointer) group_id_probe_handler);
           ^
    cc1: all warnings being treated as errors

We should cast to guintptr from gulong before we cast to gpointer.

https://bugzilla.gnome.org/show_bug.cgi?id=754755
2015-09-11 23:28:07 +02:00
Wim Taymans
3d733ad09f videoscale: fix gamma-decode option
We need to use the enum to configure the option now.
2015-09-03 17:56:05 +02:00
Thiago Santos
76d26a60bd playsinkconvertbin: implement accept-caps handler
The default one will just go through the internal elements which might
just be identity when it is in passthrough which will lead to the query
being handled by the downstream sink, ignoring all that playsinkconvertbin
could actually handle and convert.

https://bugzilla.gnome.org/show_bug.cgi?id=754235
2015-08-28 09:44:10 -03:00
Eunhae Choi
b1f78b5d23 subparse: use g_clear_error instead of g_error_free
To avoid invalid pointer accees the err pointer should be set to NULL.
By using g_clear_error() it calls free and clear the pointer.

https://bugzilla.gnome.org/show_bug.cgi?id=753817
2015-08-19 16:21:25 +03:00
Sebastian Dröge
2727ca01f5 Revert "decodebin: Handle the preroll multi-queue size"
This reverts commit 5c8ef0ea05.
2015-08-18 18:47:22 +03:00
Sebastian Dröge
4fe4357188 Revert "decodebin: Store extra_buffer_required per group, not globally"
This reverts commit 1ea81114ea.
2015-08-18 18:47:21 +03:00
Sebastian Dröge
970bc16bf8 Revert "decodebin: If extra buffers are going to be required, we're still prerolling"
This reverts commit a3b24f0241.
2015-08-18 18:47:18 +03:00
Sebastian Dröge
a3b24f0241 decodebin: If extra buffers are going to be required, we're still prerolling 2015-08-18 15:19:03 +03:00