Commit graph

4224 commits

Author SHA1 Message Date
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