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Author SHA1 Message Date
Stefan Sauer
fb162c8eb4 controller: port to new controller location and api 2011-11-04 20:15:48 +01:00
Wim Taymans
7753feb4fd pulseaudiosink: more 0.11 fixing 
Make sure the caps event gets to the sink.
 2011-11-04 16:21:13 +01:00
Wim Taymans
f6f8d9bb17 pulseaudiosink: port some more 
Rename decodebin2 -> decodebin some more
Cleanup up sinkpad event handling
 2011-11-04 15:35:42 +01:00
Wim Taymans
1352a08a71 pulseaudiosink: port some more to 0.11 
We must not forward the caps event. instead we will decide what to do when the
pad block is taken.
Use decodebin instead of decodebin2
 2011-11-04 13:56:06 +01:00
Wim Taymans
e038ab5a0b tags: update for tag API removal 2011-11-02 12:09:20 +01:00
Wim Taymans
22eb0d2300 Merge branch 'master' into 0.11 2011-11-02 10:40:12 +01:00
Wim Taymans
9a8a8e72c8 structure: fix for api update 2011-11-02 09:06:37 +01:00
Tim-Philipp Müller
9f77b02b15 Update for pad API changes 
GstProbeType, GstProbeReturn and GstActivateMode -> GstPad*
 2011-11-01 00:52:28 +00:00
Tim-Philipp Müller
d6e1f53233 flacenc: remove dead code from header 
We require a new-enough libflac that this condition will never apply.
 2011-10-30 19:30:14 +00:00
Tim-Philipp Müller
a49818f876 flacdec: parse stream headers from caps in set_format function 
Not that this seems to be actually needed, libflac happily decodes
stuff even if we just drop all headers and never feed it to the
library.
 2011-10-30 19:12:44 +00:00
Tim-Philipp Müller
ab591b6d53 flacdec: don't extract metadata, leave that to the parser or container 2011-10-30 19:12:44 +00:00
Tim-Philipp Müller
5ab43cdf91 flacdec: we expect framed input now, remove some more code 2011-10-30 19:12:39 +00:00
Tim-Philipp Müller
92361863e6 flacdec: naive port to GstAudioDecoder 
This would probably have been too invasive to do in the 0.10
branch, with all the pull-mode and parser handling code in
there.
 2011-10-30 17:39:40 +00:00
Tim-Philipp Müller
9cd17092d8 ext, gst: update for taglist API changes 2011-10-30 11:44:53 +00:00
Wim Taymans
d40e915449 Merge branch 'master' into 0.11 2011-10-28 16:52:08 +02:00
Wim Taymans
3389e79f38 pulseaudiosink: fix porting errors 
The probes were ported wrongly and caused deadlocks.
 2011-10-28 15:11:10 +02:00
Tim-Philipp Müller
ff40deb139 jpegdec: add sof-marker to template caps, so we don't get plugged for lossless jpeg 
jpegdec (using libjpeg 6.2/8) can't decode some lossless types of JPEG.

https://bugzilla.gnome.org/show_bug.cgi?id=556648
 2011-10-28 12:10:34 +01:00
Wim Taymans
65c71b0717 pulse: fix check for empty caps 2011-10-28 12:51:31 +02:00
Wim Taymans
4b6a226263 Merge branch 'master' into 0.11 
Conflicts:
	ext/pulse/pulseaudiosink.c
	ext/pulse/pulsesink.c
 2011-10-27 16:08:22 +02:00
Wim Taymans
fc4684f4c6 fix compilation 2011-10-27 16:03:17 +02:00
Stefan Sauer
2468492f02 interfaces: clean up the use of iface and class/klass 2011-10-21 14:58:41 +02:00
René Stadler
41116224c8 pngenc: increase arbitrary resolution limits 
Apparently libpng can technically do up to 2^31-1 rows and columns. However it
imposes an (arbitrary) default limit of 1 million (that could theoretically be
lifted by using some additional API).

Moved array allocation to the heap now.
 2011-10-21 10:27:04 +02:00
René Stadler
db1f10adc8 pngenc: don't unconditionally allocate 4096 pointers on the stack 
Instead allocate as many as needed (on the stack still).
 2011-10-21 10:26:48 +02:00
René Stadler
65f9354803 pngenc: ensure setcaps was called before chain function 
This is needed to properly error out for e.g. "fakesrc ! pngenc ! fakesink".
 2011-10-21 10:26:20 +02:00
René Stadler
7e390c4635 pngenc: validate input buffer size 
Just for safety; of course such mismatch represents a bug in another element.
 2011-10-21 10:25:51 +02:00
René Stadler
9eb55c3d8f pngenc: make setcaps more robust, use gstvideo functions 
A setcaps function needs to actually verify the caps carefully. In this case,
it was possible to e.g. link a video decoder with YUV+RGB template caps to
pngenc.  That would cause a crash when the decoder pushes a YUV buffer. Same
thing when pushing a valid buffer that exceeds the resolution limits.

Also, missing framerate caps field would cause a glib critical warning due to
invalid GValue. This fails hard now.
 2011-10-21 10:25:08 +02:00
Arun Raghavan
a7790efd04 pulse: Get caps correctly on pad block 
Instead of always going upstream, we should first see if already got
caps from a setcaps() call.

https://bugzilla.gnome.org/show_bug.cgi?id=661262
 2011-10-18 20:02:55 +05:30
Tim-Philipp Müller
e9ad06e202 wavpackenc: don't unref buffer with gst_object_unref() 2011-10-18 12:25:14 +01:00
Wim Taymans
6de67bb014 pulsesink: only use is_pcm for 1.0 of pulseaudio 2011-10-18 12:05:01 +02:00
Wim Taymans
0ade1a5822 pulsesink: only disable trickmodes for !pcm 
Only disable trickmodes when we are not dealing with raw PCM samples.
 2011-10-18 11:58:57 +02:00
Edward Hervey
1a10116bbe flacenc: Properly register type 
It's a subclass of GstAudioEncoder and not of GstElement
 2011-10-13 17:12:23 +02:00
Wim Taymans
16649b2508 fix compile 2011-10-13 09:02:47 +02:00
Wim Taymans
a5cc912140 Merge branch 'master' into 0.11 
Conflicts:
	ext/jpeg/gstjpegdec.c
	gst/rtp/gstrtpvrawpay.c
 2011-10-13 08:58:06 +02:00
Thiago Santos
0e167e59d4 pulseaudiosink: Use new GstIterator API correctly 
GstIterator now uses GValue, use it correctly.
 2011-10-12 07:36:09 -03:00
Sjoerd Simons
95db648516 jpegdec: Implement upstream negotiation 
Add upstream negotiation for jpegdec. Fixes #660275
 2011-10-10 21:37:10 +01:00
Wim Taymans
03ca12d974 annodex: port to 0.11 2011-10-10 12:27:06 +02:00
Wim Taymans
aea9b5e8c8 Merge branch 'master' into 0.11 
Conflicts:
	ext/speex/gstspeexenc.c
 2011-10-10 11:48:20 +02:00
Thiago Santos
b09704020c pulse: port pulseutil to 0.11 2011-10-10 00:18:56 -03:00
Thiago Santos
4517eb28c0 pulseaudiosink: port to 0.11 2011-10-09 21:19:30 -03:00
Thiago Santos
358767e217 pulsesink: Fixing getcaps function 
Update getcaps function to 0.11 API
 2011-10-09 21:19:27 -03:00
Mark Nauwelaerts
00a91fc061 speexenc: only push header buffers following initial events 2011-10-09 21:32:32 +02:00
Wim Taymans
586ef0babd Merge branch 'master' into 0.11 
Conflicts:
	ext/speex/gstspeexdec.c
	ext/speex/gstspeexenc.c
	gst/isomp4/atoms.c
	gst/isomp4/gstqtmux.c
 2011-10-06 12:23:39 +02:00
Tim-Philipp Müller
ca77c96c51 speexenc: initialise variable before adding to it 2011-09-29 23:21:46 +01:00
Mark Nauwelaerts
c5354bee04 speexdec: port to audiodecoder 2011-09-29 17:33:25 +02:00
Mark Nauwelaerts
53476c1580 speexenc: clean up some unused remnants 2011-09-29 17:33:23 +02:00
Mark Nauwelaerts
c1909c32c5 speexenc: port to audioencoder 2011-09-29 17:33:21 +02:00
Tim-Philipp Müller
3d01b9f398 flacdec: get rid of granulepos handling 
Leave that to the parser or demuxer. There's still some
code for operating in DEFAULT (samples) format, but that
will be removed later.
 2011-09-28 19:10:27 +01:00
Tim-Philipp Müller
5c28f426d7 flacdec: get rid of pull-mode support and focus on being a decoder 
Leave all the other stuff to flacparse.
 2011-09-28 19:03:13 +01:00
Tim-Philipp Müller
e0d994c9e1 flac, jpeg: fix compiler warning 2011-09-28 17:39:06 +01:00
Wim Taymans
b4524858be flac: port to 0.11 2011-09-28 17:40:01 +02:00
Wim Taymans
762602d56a Merge branch 'master' into 0.11 
Conflicts:
	ext/flac/gstflacenc.c
 2011-09-28 17:39:12 +02:00
Mark Nauwelaerts
e8bcd41d73 flacenc: port to audioencoder 2011-09-28 16:14:46 +02:00
Wim Taymans
87fbd1e784 Merge branch 'master' into 0.11 
Conflicts:
	common
	ext/pulse/pulsesink.c
	ext/soup/gstsouphttpclientsink.c
	gst/audioparsers/gstaacparse.c
	gst/audioparsers/gstac3parse.c
	gst/rtp/gstrtph264depay.c
	gst/rtpmanager/gstrtpjitterbuffer.c
	gst/rtpmanager/rtpjitterbuffer.c
	gst/rtsp/gstrtspsrc.c
	sys/ximage/gstximagesrc.c
 2011-09-28 12:44:59 +02:00
Tim-Philipp Müller
3828537857 soup: rename souphttpsink to souphttpclientsink 
To avoid confusion, and because we might want a server
sink at some point too.

https://bugzilla.gnome.org/show_bug.cgi?id=659947
 2011-09-25 15:13:39 +01:00
Tim-Philipp Müller
be7cbd4c21 souphttpsink: don't create unused second sink pad object 
The base class will create the sink pad.
 2011-09-23 16:39:46 +01:00
Vincent Penquerc'h
7e4574e968 speexenc: do not use invalid buffer timestamps 2011-09-19 09:37:58 +02:00
Arun Raghavan
8ca420f547 pulse: New pulseaudiosink element to handle format changes 
This introduces a new bin which wraps around pulsesink and depending on
the formats supported by the sink, plugs in/out a decodebin2 as
required. This allows users to switch sinks on the stream and adapts
accordingly (for example, you could watch a movie in passthrough mode on
your receiver which supports AC3 decode, then plug out and switch to a
non-digital profile to continue uninterrupted on analog output).

The bin is required because doing the same with playbin2/playsink will
require API changes that cannot be made in 0.10. With 0.11/1.0, we
should be able to ask for upstream caps renegotiation to deal with all
this.

https://bugzilla.gnome.org/show_bug.cgi?id=657179
 2011-09-19 07:43:04 +05:30
Konstantin Miller
24d002e04d souphttpsrc: Don't handle HTTP response 407 as error if proxy authentication data is available 
Fixes bug #657422.
 2011-09-07 13:28:45 +02:00
Wim Taymans
33f18b8ea4 Merge branch 'master' into 0.11 
Conflicts:
	gst/audioparsers/gstamrparse.c
	gst/isomp4/qtdemux.c
 2011-09-06 16:06:25 +02:00
Wim Taymans
e204c5934c -good: port to new audio caps 2011-09-06 13:16:27 +02:00
Sebastian Dröge
7b592ff126 souphttpsrc: Allow positive, non-1.0 segment rates 
Only negative rates are not supported. Fixes bug #658305.
 2011-09-06 10:34:35 +02:00
Wim Taymans
85d7fe14b2 soup: port soup elements to 0.11 2011-08-29 18:02:15 +02:00
Wim Taymans
34ea60526d pulse: add some more channels 2011-08-24 18:44:01 +02:00
Wim Taymans
e9df54819c Merge branch 'master' into 0.11 2011-08-24 14:16:44 +02:00
Arun Raghavan
bd604175c5 pulsesink: Trivial indentation fix 2011-08-23 22:48:34 +05:30
Monty Montgomery
799c8e3d04 flacdec: Correct sample number rounding resulting in timestamp jitter 
flacdec converts the src timestamp to a sample number, uses that internally, then reconverts the sample number to a timestamp for the output buffer.  Unfortunately, sample numbers can't be represented in an integer number of nanoseconds, and the conversion process was truncating rather than rounding, resulting in sample numbers and output timestamps that were often off by a full sample.

This corrects the time->sample convesion
 2011-08-23 10:09:41 +02:00
Wim Taymans
0eeffef222 pulsesink: port after merge 2011-08-19 16:13:23 +02:00
Wim Taymans
e1b795ac13 Merge branch 'master' into 0.11 2011-08-19 16:12:01 +02:00
Wim Taymans
77ad0a1363 port more elements to new audio caps and API 2011-08-19 14:01:45 +02:00
David Henningsson
e70020b456 pulsesink: Allow writes in bigger chunks 
There's no use in splitting the incoming data down to the segsize
limit - by writing as much as possible in one chunk, we increase
performance and avoid PulseAudio unnecessary rewinds.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
 2011-08-19 09:48:27 +02:00
Wim Taymans
09b15d7dfe port to new audio caps. 2011-08-18 19:21:07 +02:00
Wim Taymans
ce1e7cb108 Merge branch 'master' into 0.11 
Conflicts:
	ext/flac/gstflacdec.c
 2011-08-17 15:52:18 +02:00
Wim Taymans
be4f60b062 jpeg: port to 0.11 
Also disable smoke for now.
 2011-08-17 15:39:27 +02:00
Vincent Penquerc'h
3e0134f51f flacdec: avoid timestamp/offset tracking going out of sync 
The libFLAC API is callback based, and we must only call it to
output data when we know we have enough input data. For this
reason, a single processing step is done when receiving a buffer.
However, if there were metadata buffers still pending, a step
intended for the first audio frame might end up writing that
leftover metadata. Since a single step is done per buffer, this
will cause every buffer to be written one step late.

This would add some latency (a bufferfull's worth), possibly
lose a buffer when seeking or the like, and also cause timestamp
and offset to be applied to the wrong buffer, as updates to
the "current" segment last_stop (from incoming buffer timestamp)
will be applied to an output buffer originating from the previous
incoming buffer.

This fixes the issue by ensuring that, upon receiving the first
audio frame, processing is done till all metadata is processed,
so the next "single step" done will be for the audio frame. After
this, we should keep to 1 input buffer -> 1 output buffer and so
avoid getting out of sync.

https://bugzilla.gnome.org/show_bug.cgi?id=650960
 2011-08-17 13:40:59 +01:00
Vincent Penquerc'h
e09eb95a5f flacdec: bail on reserved value 
Now that we look at the right bits, we can test against the reserved
value as we do for other fields.

https://bugzilla.gnome.org/show_bug.cgi?id=650960
 2011-08-17 00:02:38 +01:00
Vincent Penquerc'h
64beef4610 flacdec: fix bit twiddling 
Right shifting a 8 bit value by 8 bits is twice too much
to get the high 4 bits.

https://bugzilla.gnome.org/show_bug.cgi?id=650960
 2011-08-17 00:01:37 +01:00
Vincent Penquerc'h
1549aaba27 flacdec: warn if we see a variable block size where unsupported 
https://bugzilla.gnome.org/show_bug.cgi?id=650960
 2011-08-17 00:01:07 +01:00
Wim Taymans
4bb2b140e9 Merge branch 'master' into 0.11 
Conflicts:
	sys/v4l2/v4l2src_calls.c
 2011-08-16 18:35:53 +02:00
Tim-Philipp Müller
26a3a12513 jackaudiosrc: fix error message code 
And also post 'not found' error if jackd is not even installed.
 2011-08-13 16:52:53 +01:00
Edward Hervey
145f6da5bb aasink: Remove unused variables 2011-08-10 11:28:26 +02:00
Tim-Philipp Müller
9f904ac438 aalib: make sure -DGST_USE_UNSTABLE_API is defined 
So we don't get warnings.
 2011-08-08 15:26:00 +01:00
Wim Taymans
71346020d5 pulsesrc: avoid race in starting 
Sine the base class now does the negotiation from the streaming thread we have
to be careful and check if the stream is ready before changing its corked state.
 2011-08-07 11:17:41 +02:00
Wim Taymans
d9750387c1 pulse: more cleanups 2011-08-04 18:41:29 +02:00
Wim Taymans
9ae85cb662 pulsesrc: small cleanups 2011-08-04 18:15:55 +02:00
Wim Taymans
fcbe26cd6f pulsesrc: small cleanups 2011-08-04 16:32:39 +02:00
Wim Taymans
ee2aa25e04 port to new API 2011-08-03 18:37:27 +02:00
Wim Taymans
4121021bb2 Merge branch 'master' into 0.11 
Conflicts:
	ext/pulse/pulsesink.c
	ext/pulse/pulsesrc.c
	gst/audioparsers/gstac3parse.c
	gst/rtp/gstrtph264depay.c
	gst/rtp/gstrtph264pay.c
	gst/rtpmanager/gstrtpssrcdemux.c
 2011-08-03 18:25:30 +02:00
Sebastian Dröge
f18eccd286 hal: Remove hal plugin 
hal is not developed anymore and nobody is using the plugin nowadays.
 2011-08-03 10:59:56 +02:00
Tristan Matthews
c26442a3ba jackaudiosink: Don't call g_alloca() in process_cb 
g_alloca() is not RT-safe, so instead we should allocate the
memory needed in advance. Fixes #655866
 2011-08-03 09:44:05 +02:00
Tim-Philipp Müller
25ace0e524 pulsesink: fix variable-set-but-not-used compiler warning with older pulse versions 2011-07-29 13:05:42 +01:00
Arun Raghavan
ac7cad431c pulsesink: Add support for compressed formats 
This adds support for various compressed formats (AC3, E-AC3, DTS and
MP3) payloaded in IEC 61937 format (used for transmission over S/PDIF,
HDMI and Bluetooth).

The acceptcaps() function allows bins to probe for what formats the sink
being connected to support. This only works after the element is set to
at least READY.

If the underlying sink changes and the format we are streaming is not
available, we emit a message that will allow upstream elements/bins to
block and renegotiate a new format.
 2011-07-29 01:25:15 +05:30
Arun Raghavan
a67b536741 pulsesink: Use the extended stream API if available 
This uses the new extended API for creating streams. This will allow us
to support compressed formats natively in pulsesink as well.
 2011-07-29 01:25:15 +05:30
Arun Raghavan
379049809c pulsesrc: Add a source-output-index property 
This exposes the source output index of the record stream that we open
so that clients can use this with the introspection if they want (to
move the stream, for example).
 2011-07-29 00:07:52 +05:30
Tim-Philipp Müller
ab62599832 speex: update for position/query/convert API changes 2011-07-28 11:38:31 +01:00
Thiago Santos
14b9fb7be6 pulsesrc: Fix default value leaking 
Remember to free the default value of client name, avoiding a
leak
 2011-07-18 15:16:01 -03:00
Wim Taymans
da28ebfbe3 aasink: port to new video API 2011-07-06 17:50:54 +02:00
Wim Taymans
1a0a6f54bb cacasink: port to 0.11 2011-07-06 17:40:20 +02:00
Wim Taymans
f70da0a542 jpeg: beginnings of porting to 0.11 2011-07-06 16:51:36 +02:00
Wim Taymans
fdf5a49422 speex: port speex elements 2011-07-06 15:57:23 +02:00
Wim Taymans
3fd1106b7e Merge branch 'master' into 0.11 2011-07-06 12:05:12 +02:00
René Stadler
ae87731de5 pulsesink: prevent race condition causing ref leak 
Since commit 8bfd80, gst_pulseringbuffer_stop doesn't wait for the
deferred call to be run before returning. This causes a race when
READY->NULL is executed shortly after, which stops the mainloop. This
leaks the element reference which is passed as userdata for the callback
(introduced in commit 7cf996, bug #614765).

The correct fix is to wait in READY->NULL for all outstanding calls to
be fired (since libpulse doesn't provide a DestroyNotify for the
userdata). We get rid of the reference passing from 7cf996 altogether,
since finalization from the callback would anyways lead to a deadlock.

Re-fixes bug #614765.
 2011-07-05 16:36:17 +02:00
René Stadler
f8456e2a1a pulsesink: small cleanup of copy-paste code 2011-07-05 16:36:17 +02:00
René Stadler
3589cee762 pulsesink: remove unused member variable and misleading log message 
Wim changed it in commit 8bfd80 so that pa_defer_ran is not read
anywhere.

The log message used to annotate a mainloop_wait call which is gone.
 2011-07-05 16:36:17 +02:00
Wim Taymans
8b040cfae2 pulse: remove implementsinterface 2011-07-04 18:12:56 +02:00
Mark Nauwelaerts
0c25863253 jpegdec: avoid crashing on invalid input without components 2011-07-04 14:32:27 +02:00
Mark Nauwelaerts
d59a00aa1c Merge branch 'master' into 0.11 
Conflicts:
	ext/pulse/pulsesink.c
 2011-07-04 11:48:13 +02:00
David Schleef
f69dcaab58 pulse: Increase ranks to PRIMARY + 10 
So that pulsesrc/pulsesink get chosen over other possible PRIMARY
src/sinks by autoaudiosink.  Presumably, if pulse is available, it
is always preferred over another src/sink.

Fixes: #647540.
 2011-07-03 19:53:42 -07:00
David Schleef
2f94df8032 jpegenc: Don't round up size of encoded buffers 
For some reason, in code dating to 2001, encoded jpeg buffers were
rounded up to multiples of 4 bytes.  With the added bonus that the
extra bytes are unwritten, causing valgrind issues.  Oops.  I can't
think of any reason why JPEG buffers need to be multiples of 4 bytes,
so I removed the padding.  There might be some code somewhere that
depends on this behavior, so if this needs to be reverted, please fix
the valgrind issues.
 2011-06-29 23:55:33 -07:00
Andoni Morales Alastruey
d9f4c59c49 dv1394src: make the internal clock thread safe 
Fixes: #653091.
 2011-06-24 12:01:39 -07:00
Arun Raghavan
04786a6d31 pulse: Drop support for PA versions before 0.9.16 
This drops support fof PulseAudio versions prior to 0.9.16, which was
released about 1.5 years ago. Testing with very old versions is not
feasible and we don't want to maintain 2 independent code-paths.
 2011-06-21 16:47:11 -07:00
Wim Taymans
adee56222b pulsesink: fix for header cleanups 2011-06-21 18:19:02 +02:00
Tim-Philipp Müller
3d6955d686 Remove esound/esdsink plugin 2011-06-15 15:09:18 +01:00
Tim-Philipp Müller
728732438b Remove gconf elements and plugin 
GConf was deprecated in favour of GSettings etc.
 2011-06-15 14:37:29 +01:00
Wim Taymans
c2358404c1 jack: port some more 2011-06-09 17:56:18 +02:00
Wim Taymans
3446ca801e aasink: register template and klass correctly 2011-06-09 11:37:47 +02:00
Wim Taymans
710fa239d5 Merge branch 'master' into 0.11 2011-06-08 18:06:56 +02:00
Wim Taymans
0af32751ae assink: port aasink to 0.11 2011-06-08 18:05:20 +02:00
David Schleef
6d8d08eac7 souphttpsink: code cleanup 2011-06-04 13:43:28 -07:00
Wim Taymans
0b1bdcf7cb Merge branch 'master' into 0.11 
Conflicts:
	sys/ximage/ximageutil.c
 2011-06-02 18:51:29 +02:00
Sebastian Dröge
252c1afcc2 dvdemux: First query the peer duration in the requested format before converting to BYTES 
Fixes usage of dvdemux after another demuxer, e.g. mxfdemux.
Fixes bug #650503.
 2011-06-02 13:38:30 +02:00
Sebastian Dröge
e376913fc3 souphttpsink: Fix refcounting of the "session" property 
Properties should never take ownership of the values
passed to them.
 2011-06-02 10:41:52 +02:00
David Schleef
3937abf999 soup: Add souphttpsink 2011-06-01 17:19:02 -07:00
Vincent Penquerc'h
38b596337f cairotextoverlay: forward new segment events from the sink to the source 
Not doing so will cause buffers to be received by downstream without
a time base set.
We use the same method avimux uses to get access to the event when
collectpads got the sink event function.

https://bugzilla.gnome.org/show_bug.cgi?id=640323
 2011-05-26 11:35:01 +02:00
Vincent Penquerc'h
b99f4be3db textoverlay: forward source events to sinks 
Events are passed to the video sink, and to the text sink if it is
linked.
This will allow seeking, for instance.

https://bugzilla.gnome.org/show_bug.cgi?id=586450
 2011-05-26 11:33:10 +02:00
Wim Taymans
d89790d545 Merge branch 'master' into 0.11 
Conflicts:
	gst/avi/gstavidemux.c
	gst/rtp/gstrtpac3depay.c
	gst/rtp/gstrtpg726depay.c
	gst/rtp/gstrtpmpvdepay.c
	gst/videofilter/gstgamma.c
 2011-05-24 17:34:19 +02:00
Tim-Philipp Müller
900b2fb63d flacdec: also try upstream first for duration query in DEFAULT format 
https://bugzilla.gnome.org/show_bug.cgi?id=650785
 2011-05-23 13:25:44 +01:00
Wim Taymans
cf1248e566 Merge branch 'master' into 0.11 
Conflicts:
	configure.ac
 2011-05-16 17:52:11 +02:00
Wim Taymans
a0e301ab1a -good: fix for new API 2011-05-16 17:50:15 +02:00
Pino Toscano
b52c7ebd19 pulse: Define PATH_MAX if it isn't defined 
GNU Hurd for example doesn't define it.
 2011-05-16 09:04:31 +02:00
Sebastian Dröge
d4b9cbc1ae flacdec: Send EOS when seeking after the end of file instead of failing 
Fixes bug #649780.
 2011-05-14 11:52:13 +02:00
Wim Taymans
992e3f7c55 Merge branch 'master' into 0.11 
Conflicts:
	configure.ac
	gst/rtp/gstrtpgstpay.c
 2011-04-29 15:49:41 +02:00
Wim Taymans
5ed18ad7b9 jack: port jack elements 2011-04-26 15:42:47 +02:00
Philip Jägenstedt
660987c6d7 jpegdec: documentation typo "jpegddec" 
https://bugzilla.gnome.org/show_bug.cgi?id=648589
 2011-04-25 21:36:28 +01:00
Wim Taymans
a1894ed363 Merge branch 'master' into 0.11 2011-04-25 11:38:28 +02:00
David Schleef
24d57ac961 pngdec: Remove temporary code 
Now that we depend on (what will be) -base-0.10.33.
 2011-04-24 18:53:50 -07:00
Wim Taymans
2f9a7b1a1b flac: port to 0.11 2011-04-19 18:55:31 +02:00
Wim Taymans
dea50d0b98 use G_DEFINE_TYPE 2011-04-19 17:07:18 +02:00
Wim Taymans
7555d0949f Merge branch 'master' into 0.11 
Conflicts:
	android/apetag.mk
	android/avi.mk
	android/flv.mk
	android/icydemux.mk
	android/id3demux.mk
	android/qtdemux.mk
	android/rtp.mk
	android/rtpmanager.mk
	android/rtsp.mk
	android/soup.mk
	android/udp.mk
	android/wavenc.mk
	android/wavparse.mk
	configure.ac
 2011-04-18 10:23:45 +02:00
Tim-Philipp Müller
f325935314 pulse, speexenc, rtpgsmpay: don't use g_assert() for error handling 
Don't use g_assert() for error handling, even if they're highly unlikely.
Either we *know* that something can't happen, in which case we
should just not handle it, or we think something can happen, but it is
very very unlikely that it will ever happen, in which case we should
handle it like any other error instead of asserting.

g_assert() is best left for conditions we have control of, like checking
internal consistency of our code, not checking return values of external
code.

Fixes a bunch of warnings when compiling with -DG_DISABLE_ASSERT:
gstrtpgsmpay.c: In function 'gst_rtp_gsm_pay_handle_buffer':
gstrtpgsmpay.c:130:17: warning: variable 'rtpgsmpay' set but not used
gstspeexenc.c: In function 'gst_speex_enc_encode':
gstspeexenc.c:904:19: warning: variable 'written' set but not used
pulsesink.c: In function 'gst_pulsesink_change_state':
pulsesink.c:2725:9: warning: variable 'res' set but not used
pulsesrc.c: In function 'gst_pulsesrc_change_state':
pulsesrc.c:1253:7: warning: variable 'e' set but not used
 2011-04-16 18:15:43 +01:00
Tim-Philipp Müller
a068133ee7 jack: fix unused-but-set-variable warnings with gcc-4.6 2011-04-16 13:33:45 +01:00
Robert Swain
098cf89e34 jpegenc: pulsesink: raw1394: Address unused but set variables 
GCC 4.6.x spits warnings about such usage of variables. The variables in
raw1394 were marked with G_GNUC_UNUSED as this seemed omre appropriate.
The others were removed.
 2011-04-16 13:13:04 +01:00
Sebastian Dröge
18eee7719d flacenc: Properly interprete the result of strcmp() 2011-04-13 16:17:41 +02:00
Sebastian Dröge
16f82e46f5 flacenc: Don't store image tags inside the vorbiscomments and the flac metadata 
Instead only store them inside the flac metadata. There's
no point in storing them twice and the flac metadata is
still the official way to store image tags inside flac.
 2011-04-13 16:11:44 +02:00
Stefan Kost
21a688ce25 jack: also handle deprecations for jack 1.9.7 
Jack 1.9.7 was released 20.Mar.2011, need to handle the deprecated api for this
version too.
 2011-04-11 12:44:19 +03:00
Sebastian Dröge
b784173e4a jpegdec: Unref event if the parent element disappeared 2011-04-08 15:22:47 +02:00
Havard Graff
9386448649 jpegdec: Make upstream events MT-safe 2011-04-08 15:21:52 +02:00
Sebastian Dröge
11bcac7c90 Revert "Pulsesink: Allow chunks up to bufsize instead of segsize" 
This reverts commit 1e2c1467ae.

The commit causes pulsesink to ignore the latency-time baseaudiosink property.
 2011-04-08 14:35:04 +02:00
Alexey Fisher
0016ceaa2b speexenc: Use speex intern silence detection 
Speex has build in silence detection. If speex_encode_int returns 0,
than there is silence and sample do not need to be transmitted.
This work only if vbr=1 and dtx=1 optionas are enabled.
So if we get 0, we add GAP flag to the sample.
 2011-04-08 13:54:49 +02:00
Wim Taymans
c124ba1489 Merge branch 'master' into 0.11 
Conflicts:
	gst/rtsp/gstrtspsrc.c
 2011-04-05 17:20:08 +02:00
Jon Nordby
d68dd46084 jack: Fix build with jack 0.120.1 
9544622674 checked
for 0.120.2 and later, but the deprecation was introduced in
0.120.1
 2011-04-05 13:12:28 +03:00
Arun Raghavan
dc48eaac13 pulsesink: Always call pa_stream_new_with_proplist() 
pa_stream_new_with_proplist() can take a NULL proplist, so we don't need
to concern ourselves with whether it's NULL or not.
 2011-04-04 17:23:21 +05:30
Sebastian Dröge
6fd1546bce flacenc: Add support for writing METADATA_BLOCK_PICTURE blocks for GST_TAG_IMAGE and GST_TAG_PREVIEW_IMAGE 2011-04-01 13:18:55 +02:00
Mark Nauwelaerts
176b8ffbff pulsesink: also uncork during EOS waiting (and after EOS is rendered) 
Pulsesink was recently changed to defer uncorking until there is data
to write. This condition will however never occur when EOS in being
rendered (since that marks the end of data). Changing to PAUSED state
while EOS is being waited on results in a hang: pausing corks the
stream, which will never be undone since there is no more data when
going back to PLAYING. If pulsesink is the clock provider, deadlock
ensues since time doesn't continue in corked state and the clock id
for EOS wait never fires.

Fixes #645961.
 2011-03-31 13:25:19 +02:00
Tim-Philipp Müller
c365fbddba flacdec: fix issues with large metadata blocks when streaming unframed flac 
Parse metadata blocks when handling unparsed flac in push mode. This
works around a bunch of issues with the flac decoder when handling
metadata blocks that are larger than the max. flac framesize, which
coverart blocks often are. We need to have all the data for these
blocks available when we pass data to libflac.

http://gstreamer-devel.966125.n4.nabble.com/Flac-files-that-will-playback-but-not-stream-td3338198.html#a3395276

https://bugzilla.gnome.org/show_bug.cgi?id=566769
 2011-03-28 23:46:47 +01:00
Wim Taymans
8f22a09dc4 Merge branch 'master' into 0.11-fdo 2011-03-28 20:50:59 +02:00
Stefan Kost
ed77b14aa0 cairo: fix the name of the *-marshall.list file to unbreak make distcheck 2011-03-25 12:53:43 +02:00
Sebastian Dröge
85ace6d413 speexdec: Get and use streamheader from the caps if possible 
This allows playback of streams where the streamheader buffers
were dropped from the stream for some reason.
 2011-03-24 09:00:32 +01:00
Robert Swain
a430042f41 jackaudiosink: Fix typo from 9544622674 2011-03-16 10:43:47 +01:00
Vincent Penquerc'h
56edbd0fa3 speexdec: silence warning message when appropriate 
If we did not know how many frames to expect, then we get an unexpected
end of stream when trying to decode more frames that are there, if there
are leftover bits to pad to the next byte
 2011-03-15 19:44:03 +01:00
Sebastian Dröge
4d7b4ca2ae speexdec: Always process the number of frames per packet as specified in the header 
Looking at the remaining bits in the bitstream after decoding a
single frame can't be used as loop condition. The remaining
bits might not give a complete frame and the speex decoder will
then output nothing but access uninitialized memory, which leads
to valgrind warnings.

Fixes bug #644669.
 2011-03-14 19:31:33 +01:00
Philip Jägenstedt
84decd731f pulsesink: Better fix for deadlock on failed connect 
This reverts the previous fix that would cause a double-unlock when the
stream connect failed.

https://bugzilla.gnome.org/show_bug.cgi?id=644510
 2011-03-12 00:48:03 +05:30
Arun Raghavan
387f533a07 pulsesink: Fix deadlock if connecting to PA fails 
Commit dd4ec22e introduced a deadlock in the failure path while trying
to connect to PulseAudio. This makes sure we drop the lock on the
resource mutex to avoid this.

https://bugzilla.gnome.org/show_bug.cgi?id=644510
 2011-03-11 23:19:34 +05:30
Sebastian Dröge
9e44fff3db dvdemux: Chain up to the parent class' ::send_event for non-seek events 2011-03-11 13:48:33 +01:00
Sebastian Dröge
bc6c1bbbab dvdemux: Fix refcount issues with the seek event 
Fixes bug #642963.
 2011-03-11 13:48:32 +01:00
Tim-Philipp Müller
14e1d9de3d docs: fix pulsesink gtk-doc markup 2011-03-11 09:54:36 +00:00
Philippe Normand
9544622674 jack: fix build against jack 0.120.2 
jack_port_get_total_latency() has been deprecated in favor of
jack_port_get_latency_range().

https://bugzilla.gnome.org/show_bug.cgi?id=644477
 2011-03-11 09:48:36 +00:00
Wim Taymans
338c300f5f Merge branch 'master' into 0.11 2011-03-04 16:30:36 +01:00
Stefan Kost
89914439b7 pulsesink: add a doc example for setting stream-properties 2011-03-04 15:50:49 +02:00
Stefan Kost
edfa99ef31 pulsesink: fix the xml in the docs 2011-03-04 15:50:49 +02:00
Sebastian Dröge
fdbdbfefd2 cairooverlay: Some minor cleanup 2011-03-02 23:14:36 +01:00
Jon Nordby
32dff9df75 cairooverlay: Add generic Cairo overlay video element. 
Allows applications to connect to the "draw" signal of
the element and do their custom drawing there.
Includes an example application demonstrating usage.

Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=595520
 2011-03-02 22:55:52 +01:00
Wim Taymans
db5aa9067b Merge branch 'master' into 0.11 2011-02-28 13:29:47 +01:00
Wim Taymans
59c825a938 id3: use boxed type instead of miniobject 2011-02-28 11:58:48 +01:00
Wim Taymans
c899229c48 jpegdec: add duration when extimating QoS time 
When we need to decide on the next QoS time, take into account the duration of
the buffers.
 2011-02-28 11:58:48 +01:00
Philip Jägenstedt
dd4ec22e7e pulsesink: release pa_shared_resource_mutex before pa_threaded_mainloop_wait 
Not doing so can result in a deadlock when two threads enter
gst_pulseringbuffer_open_device at the same time, as
pa_threaded_mainloop_wait releases the mainloop lock while waiting,
allowing another thread to take it, resulting in a deadlock as two
threads waits for the lock the other is holding.

https://bugzilla.gnome.org/show_bug.cgi?id=643087
 2011-02-28 16:20:22 +05:30
Philip Jägenstedt
683c96b37a pulsesink: s/ressource/resource/ 
https://bugzilla.gnome.org/show_bug.cgi?id=643087
 2011-02-28 16:20:20 +05:30
Carsten Kroll
d4eaff5e88 dvdemux: First try if upstream handles TIME seeks before handling them here 
Fixes bug #642963.
 2011-02-24 13:57:01 +01:00
Edward Hervey
6e1c701502 Revert "Check that collectpads exists before removing pad" 
This reverts commit 8e6b876e76.

Depends on a core commit that was reverted
 2011-02-21 12:25:19 +01:00
David Schleef
8e6b876e76 Check that collectpads exists before removing pad 
The core now calls release pad from finalize, at which point
the collectpads might have already been freed.
 2011-02-20 23:46:01 -08:00
David Schleef
13d441315f pngdec: Handle 16-bit-per-channel images 2011-02-20 12:12:51 -08:00
Andoni Morales Alastruey
b2fb4edcb7 souphttpsrc: Add URI query handler 
Fixes bug #642337.
 2011-02-15 09:44:01 +01:00
David Henningsson
1e2c1467ae Pulsesink: Allow chunks up to bufsize instead of segsize 
By allowing larger chunks to be sent, PulseAudio will have a
lower CPU usage. This is especially important on low-end machines,
where PulseAudio can crash if packets are coming in at a higher
rate than PulseAudio can process them.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
 2011-01-31 17:15:05 +01:00
Arun Raghavan
e18e25a324 pulsesink: Uncork stream while flushing the ringbuffer 
After starting the ringbuffer, we wait for enough data to arrive before
uncorking the stream. This will cause the pipeline to stall if we get an
EOS (or otherwise need to flush the stream) before sufficient data
becomes available. This patch makes sure that the stream is uncorked
while flushing to avoid this problem.

Fixes issue with a webkit unit test testing reverse playback of
an MP4 H.264/AAC file.

https://bugzilla.gnome.org/show_bug.cgi?id=639740
 2011-01-17 15:40:12 +00:00
Arun Raghavan
7280ea1f68 pulsesink: Make corking during pause synchronous 
This makes the call to pa_stream_cork() during ringbuffer pause()
synchronous, which makes sure that the clock does not advance after we
take a snapshot for start_time.

https://bugzilla.gnome.org/show_bug.cgi?id=639240
 2011-01-11 20:43:54 +00:00
Wim Taymans
c17f7a943d cacasink: fix masks and strides 
Use the right endianness to read the masks.
Use the right strides for the bitmap.

Fixes #638569
 2011-01-03 18:06:48 +01:00
Tim-Philipp Müller
57e35559c8 jack: fix up element details and some other minor clean-ups 2011-01-02 19:51:54 +00:00
Tim-Philipp Müller
79a8e0ee42 jack: template caps had lists with one value, just use value directly 2011-01-02 17:40:41 +00:00
Tim-Philipp Müller
1f5083a17c jack: make get_type functions thread-safe 
Because we can (shouldn't be needed with other workarounds still there).
 2011-01-02 17:27:35 +00:00
Tim-Philipp Müller
67f4489cd0 jack: new jackaudiosrc and jackaudiosink elements, moved from gst-plugins-bad 
https://bugzilla.gnome.org/show_bug.cgi?id=621929
 2011-01-02 15:26:30 +00:00
Stefan Kost
3c43a9e102 various (ext): add missing G_PARAM_STATIC_STRINGS flags 
Canonicalize property names as needed.
 2011-01-02 14:30:09 +00:00
Tristan Matthews
11c4157af4 jack: added translatable text for server not found error 2011-01-02 14:30:08 +00:00
Tristan Matthews
c070cfa7c8 jack: added client property 2011-01-02 14:30:08 +00:00
Tristan Matthews
a110fe3343 jack: removed unused file gstjackbin.c 
This is a 0.8 leftover.
 2011-01-02 14:30:08 +00:00
Wim Taymans
71b06e4957 jacksrc: make sure we always read nframes 
Error out when we are asked to read a different size that what was configured as
the jack period size because that would mean something else is wrong.

Fixes #618409
 2011-01-02 14:30:08 +00:00
Tristan Matthews
e05b723274 jack: improve process_cb 2011-01-02 14:30:08 +00:00
Tristan Matthews
55f1ea09b0 jack: implement multichannel support correctly for jackaudiosrc 
Fixes parts of bug #616541.
 2011-01-02 14:30:08 +00:00
Stefan Kost
3153aa995f jack: remove empty dispose and finalize methods 2011-01-02 14:30:08 +00:00
Stefan Kost
2b7ca1413a jack: don't leak caps 
Add dispose methods to clear caps.
 2011-01-02 14:30:08 +00:00
Stefan Kost
084e8269c2 jack: don't use GST_DEBUG_FUNCPTR for gobject vmethods 2011-01-02 14:30:08 +00:00
Stefan Kost
0a22a42268 jack: fix element name in section doc blob 2011-01-02 14:30:08 +00:00
Benjamin Otte
de91f84dfb Add -Wold-style-definition 
and fix the warnings
 2011-01-02 14:30:08 +00:00
Benjamin Otte
71366f80d9 Add -Wmissing-declarations -Wmissing-prototypes to configure flags 
And fix all warnings
 2011-01-02 14:30:07 +00:00
Benjamin Otte
4ccda26327 gst_element_class_set_details => gst_element_class_set_details_simple 2011-01-02 14:30:07 +00:00
Stefan Kost
f20c73684b jack: ensure segtotal is at least 2 
Not only adjust buffer-time and avoid segtotal=0, but instead ensure segtotal is
atleast 2. Do same change on jacksrc. We could also check the latency and buffer
time configured by the client and adjust buffer-time so that we get to the same
number of segments.
 2011-01-02 14:30:07 +00:00
Stefan Kost
45fb781f9b jack: don't crash in ringbuffer with SIGFPE on small buffer-times 
Jack overrides user-specified latency-time with the one it gets from jack
itself. It also needs to adjust buffer-time somewhat to avoid segtotal being 0
 2011-01-02 14:30:07 +00:00
Stefan Kost
8a002a1121 jack: when stopping playback, do one more cycle to flush the port. Fixes #582167 
The gst_jack_audio_client_set_active() flags the port as deactivating and uses
a GCond to wait until the jack_process_cb() has run once more and cleared the
flag. This way the client zero's the buffer. This happens if one manyally go
to PAUSED and then to READY, while leting the mainloop run inbetween.
 2011-01-02 14:30:07 +00:00
Wim Taymans
7dde7eca28 jack: Add new connection mode 
Add a new connection mode to jacksrc and jacksink. In this new auto-force
connection mode jack will create as many ports as requested/needed in the
pipeline and will then connect as many physical ports as possible, possibly
leaving some ports unconnected.

Also get rid of some leftover g_print.

Fixes #575284.
 2011-01-02 14:30:07 +00:00
Stefan Kost
fb03eb6bac ext/jack/: Query port latencies for sink/src delays. 
Original commit message from CVS:
* ext/jack/gstjackaudiosink.c:
* ext/jack/gstjackaudiosrc.c:
Query port latencies for sink/src delays.
* ext/jack/gstjackbin.c:
No printf please.
 2011-01-02 14:30:07 +00:00
Stefan Kost
7ba1bcb6aa Don't install static libs for plugins. Fixes #550851 for -bad. 
Original commit message from CVS:
* ext/alsaspdif/Makefile.am:
* ext/amrwb/Makefile.am:
* ext/apexsink/Makefile.am:
* ext/arts/Makefile.am:
* ext/artsd/Makefile.am:
* ext/audiofile/Makefile.am:
* ext/audioresample/Makefile.am:
* ext/bz2/Makefile.am:
* ext/cdaudio/Makefile.am:
* ext/celt/Makefile.am:
* ext/dc1394/Makefile.am:
* ext/dirac/Makefile.am:
* ext/directfb/Makefile.am:
* ext/divx/Makefile.am:
* ext/dts/Makefile.am:
* ext/faac/Makefile.am:
* ext/faad/Makefile.am:
* ext/gsm/Makefile.am:
* ext/hermes/Makefile.am:
* ext/ivorbis/Makefile.am:
* ext/jack/Makefile.am:
* ext/jp2k/Makefile.am:
* ext/ladspa/Makefile.am:
* ext/lcs/Makefile.am:
* ext/libfame/Makefile.am:
* ext/libmms/Makefile.am:
* ext/metadata/Makefile.am:
* ext/mpeg2enc/Makefile.am:
* ext/mplex/Makefile.am:
* ext/musepack/Makefile.am:
* ext/musicbrainz/Makefile.am:
* ext/mythtv/Makefile.am:
* ext/nas/Makefile.am:
* ext/neon/Makefile.am:
* ext/ofa/Makefile.am:
* ext/polyp/Makefile.am:
* ext/resindvd/Makefile.am:
* ext/sdl/Makefile.am:
* ext/shout/Makefile.am:
* ext/snapshot/Makefile.am:
* ext/sndfile/Makefile.am:
* ext/soundtouch/Makefile.am:
* ext/spc/Makefile.am:
* ext/swfdec/Makefile.am:
* ext/tarkin/Makefile.am:
* ext/theora/Makefile.am:
* ext/timidity/Makefile.am:
* ext/twolame/Makefile.am:
* ext/x264/Makefile.am:
* ext/xine/Makefile.am:
* ext/xvid/Makefile.am:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/dshow/Makefile.am:
* gst/aiffparse/Makefile.am:
* gst/app/Makefile.am:
* gst/audiobuffer/Makefile.am:
* gst/bayer/Makefile.am:
* gst/cdxaparse/Makefile.am:
* gst/chart/Makefile.am:
* gst/colorspace/Makefile.am:
* gst/dccp/Makefile.am:
* gst/deinterlace/Makefile.am:
* gst/deinterlace2/Makefile.am:
* gst/dvdspu/Makefile.am:
* gst/festival/Makefile.am:
* gst/filter/Makefile.am:
* gst/flacparse/Makefile.am:
* gst/flv/Makefile.am:
* gst/games/Makefile.am:
* gst/h264parse/Makefile.am:
* gst/librfb/Makefile.am:
* gst/mixmatrix/Makefile.am:
* gst/modplug/Makefile.am:
* gst/mpeg1sys/Makefile.am:
* gst/mpeg4videoparse/Makefile.am:
* gst/mpegdemux/Makefile.am:
* gst/mpegtsmux/Makefile.am:
* gst/mpegvideoparse/Makefile.am:
* gst/mve/Makefile.am:
* gst/nsf/Makefile.am:
* gst/nuvdemux/Makefile.am:
* gst/overlay/Makefile.am:
* gst/passthrough/Makefile.am:
* gst/pcapparse/Makefile.am:
* gst/playondemand/Makefile.am:
* gst/rawparse/Makefile.am:
* gst/real/Makefile.am:
* gst/rtjpeg/Makefile.am:
* gst/rtpmanager/Makefile.am:
* gst/scaletempo/Makefile.am:
* gst/sdp/Makefile.am:
* gst/selector/Makefile.am:
* gst/smooth/Makefile.am:
* gst/smoothwave/Makefile.am:
* gst/speed/Makefile.am:
* gst/speexresample/Makefile.am:
* gst/stereo/Makefile.am:
* gst/subenc/Makefile.am:
* gst/tta/Makefile.am:
* gst/vbidec/Makefile.am:
* gst/videodrop/Makefile.am:
* gst/videosignal/Makefile.am:
* gst/virtualdub/Makefile.am:
* gst/vmnc/Makefile.am:
* gst/y4m/Makefile.am:
* sys/acmenc/Makefile.am:
* sys/cdrom/Makefile.am:
* sys/dshowdecwrapper/Makefile.am:
* sys/dshowsrcwrapper/Makefile.am:
* sys/dvb/Makefile.am:
* sys/dxr3/Makefile.am:
* sys/fbdev/Makefile.am:
* sys/oss4/Makefile.am:
* sys/qcam/Makefile.am:
* sys/qtwrapper/Makefile.am:
* sys/vcd/Makefile.am:
* sys/wininet/Makefile.am:
* win32/common/config.h:
Don't install static libs for plugins. Fixes #550851 for -bad.
 2011-01-02 14:30:07 +00:00
Jan Schmidt
051dba6c01 Fix compiler warnings on OS/X 
Original commit message from CVS:
* ext/jack/gstjackaudiosink.c: (jack_process_cb):
* gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew):
Fix compiler warnings on OS/X
 2011-01-02 14:30:07 +00:00
Stefan Kost
e0d25d42db ext/jack/gstjackaudiosrc.c: Try committing this once again. Now properly renamed. 
Original commit message from CVS:
* ext/jack/gstjackaudiosrc.c:
Try committing this once again. Now properly renamed.
 2011-01-02 14:30:07 +00:00
Stefan Kost
37f51bb59d docs/plugins/: docs/plugins/inspect/plugin-jack.xml 
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
* docs/plugins/gst-plugins-bad-plugins.interfaces:
* docs/plugins/gst-plugins-bad-plugins.prerequisites:
* docs/plugins/inspect/plugin-jack.xml
Add new element to docs.
* ext/jack/gstjack.h
Add missing file.
* ext/jack/gstjackaudiosrc.c:
* ext/jack/gstjackaudiosrc.h:
Rename jackaudiosrc to jack_audio_src.
 2011-01-02 14:30:06 +00:00
Tristan Matthews
b34bd1425c ext/jack/: Add a jackaudiosrc. Refactor sink slightly for better code reuse. 
Original commit message from CVS:
patch by: Tristan Matthews <tristan@sat.qc.ca>
* ext/jack/Makefile.am:
* ext/jack/gstjack.c:
* ext/jack/gstjackaudioclient.c:
* ext/jack/gstjackaudiosink.c:
* ext/jack/gstjackaudiosink.h:
* ext/jack/gstjackaudiosrc.c:
* ext/jack/gstjackaudiosrc.h:
* ext/jack/gstjackringbuffer.h:
Add a jackaudiosrc. Refactor sink slightly for better code reuse.
Fixes #545197.
 2011-01-02 14:30:06 +00:00
Stefan Kost
d32bf24142 docs/plugins/: docs/plugins/inspect/plugin-mythtv.xml 
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
* docs/plugins/gst-plugins-bad-plugins.interfaces:
* docs/plugins/gst-plugins-bad-plugins.prerequisites:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-alsaspdif.xml:
* docs/plugins/inspect/plugin-amrwb.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-bayer.xml:
* docs/plugins/inspect/plugin-bz2.xml:
* docs/plugins/inspect/plugin-cdaudio.xml:
* docs/plugins/inspect/plugin-cdxaparse.xml:
* docs/plugins/inspect/plugin-dtsdec.xml:
* docs/plugins/inspect/plugin-dvb.xml:
* docs/plugins/inspect/plugin-dvdspu.xml:
* docs/plugins/inspect/plugin-faac.xml:
* docs/plugins/inspect/plugin-faad.xml:
* docs/plugins/inspect/plugin-fbdevsink.xml:
* docs/plugins/inspect/plugin-festival.xml:
* docs/plugins/inspect/plugin-filter.xml:
* docs/plugins/inspect/plugin-flvdemux.xml:
* docs/plugins/inspect/plugin-freeze.xml:
* docs/plugins/inspect/plugin-gsm.xml:
* docs/plugins/inspect/plugin-gstinterlace.xml:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
* docs/plugins/inspect/plugin-h264parse.xml:
* docs/plugins/inspect/plugin-interleave.xml:
* docs/plugins/inspect/plugin-jack.xml:
* docs/plugins/inspect/plugin-ladspa.xml:
* docs/plugins/inspect/plugin-metadata.xml:
* docs/plugins/inspect/plugin-mms.xml:
* docs/plugins/inspect/plugin-modplug.xml:
* docs/plugins/inspect/plugin-mpeg2enc.xml:
* docs/plugins/inspect/plugin-mpeg4videoparse.xml:
* docs/plugins/inspect/plugin-mpegtsparse.xml:
* docs/plugins/inspect/plugin-mpegvideoparse.xml:
* docs/plugins/inspect/plugin-musepack.xml:
* docs/plugins/inspect/plugin-musicbrainz.xml:
* docs/plugins/inspect/plugin-mve.xml:
* docs/plugins/inspect/plugin-mythtv.xml
* docs/plugins/inspect/plugin-nas.xml:
* docs/plugins/inspect/plugin-neon.xml:
* docs/plugins/inspect/plugin-nsfdec.xml:
* docs/plugins/inspect/plugin-nuvdemux.xml:
* docs/plugins/inspect/plugin-oss4.xml
* docs/plugins/inspect/plugin-rawparse.xml:
* docs/plugins/inspect/plugin-real.xml:
* docs/plugins/inspect/plugin-replaygain.xml:
* docs/plugins/inspect/plugin-rfbsrc.xml:
* docs/plugins/inspect/plugin-sdl.xml:
* docs/plugins/inspect/plugin-sdp.xml:
* docs/plugins/inspect/plugin-selector.xml:
* docs/plugins/inspect/plugin-sndfile.xml:
* docs/plugins/inspect/plugin-soundtouch.xml:
* docs/plugins/inspect/plugin-spcdec.xml:
* docs/plugins/inspect/plugin-speed.xml:
* docs/plugins/inspect/plugin-speexresample.xml:
* docs/plugins/inspect/plugin-stereo.xml:
* docs/plugins/inspect/plugin-subenc.xml
* docs/plugins/inspect/plugin-timidity.xml:
* docs/plugins/inspect/plugin-tta.xml:
* docs/plugins/inspect/plugin-vcdsrc.xml:
* docs/plugins/inspect/plugin-videosignal.xml:
* docs/plugins/inspect/plugin-vmnc.xml:
* docs/plugins/inspect/plugin-wildmidi.xml:
* docs/plugins/inspect/plugin-x264.xml:
* docs/plugins/inspect/plugin-xvid.xml:
* docs/plugins/inspect/plugin-y4menc.xml:
* ext/amrwb/gstamrwbdec.c:
* ext/amrwb/gstamrwbenc.c:
* ext/amrwb/gstamrwbparse.c:
* ext/dc1394/gstdc1394.c:
* ext/directfb/dfbvideosink.c:
* ext/ivorbis/vorbisdec.c:
* ext/jack/gstjackaudiosink.c:
* ext/mpeg2enc/gstmpeg2enc.cc:
* ext/mplex/gstmplex.cc:
* ext/musicbrainz/gsttrm.c:
* ext/mythtv/gstmythtvsrc.c:
* ext/theora/theoradec.c:
* ext/timidity/gsttimidity.c:
* ext/timidity/gstwildmidi.c:
* gst-libs/gst/app/gstappsink.c:
* gst/deinterlace/gstdeinterlace.c:
* gst/dvdspu/gstdvdspu.c:
* gst/festival/gstfestival.c:
* gst/freeze/gstfreeze.c:
* gst/interleave/deinterleave.c:
* gst/interleave/interleave.c:
* gst/modplug/gstmodplug.cc:
* gst/nuvdemux/gstnuvdemux.c:
Add missing elements to docs. Fix doc-markup: use convinience syntax
for examples (produces valid docbook), add several refsec2 when we
have several titles. Fix some types.
 2011-01-02 14:30:06 +00:00
Stefan Kost
8909b106f0 Do not use short_description in section docs for elements. We extract them from element details and there will be war... 
Original commit message from CVS:
* ext/dc1394/gstdc1394.c:
* ext/ivorbis/vorbisdec.c:
* ext/jack/gstjackaudiosink.c:
* ext/metadata/gstmetadatademux.c:
* ext/mythtv/gstmythtvsrc.c:
* ext/theora/theoradec.c:
* gst-libs/gst/app/gstappsink.c:
* gst/bayer/gstbayer2rgb.c:
* gst/deinterlace/gstdeinterlace.c:
* gst/rawparse/gstaudioparse.c:
* gst/rawparse/gstvideoparse.c:
* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/gstrtpssrcdemux.c:
* gst/selector/gstinputselector.c:
* gst/selector/gstoutputselector.c:
* gst/videosignal/gstvideoanalyse.c:
* gst/videosignal/gstvideodetect.c:
* gst/videosignal/gstvideomark.c:
* sys/oss4/oss4-mixer.c:
* sys/oss4/oss4-sink.c:
* sys/oss4/oss4-source.c:
Do not use short_description in section docs for elements. We extract
them from element details and there will be warnings if they differ.
Also fixing up the ChangeLog order.
 2011-01-02 14:30:06 +00:00
Wim Taymans
102b2e3fc4 ext/jack/gstjackaudiosink.c: Include the element name in the port name to avoid duplicate port names. 
Original commit message from CVS:
* ext/jack/gstjackaudiosink.c:
(gst_jack_audio_sink_allocate_channels):
Include the element name in the port name to avoid duplicate port names.
 2011-01-02 14:30:06 +00:00
Tim-Philipp Müller
d418db9aed ext/jack/gstjackaudiosink.c: Work around missing bits of thread-safety on older GLibs some more to avoid assertions w... 
Original commit message from CVS:
* ext/jack/gstjackaudiosink.c: (gst_jack_audio_sink_class_init):
Work around missing bits of thread-safety on older GLibs some
more to avoid assertions when starting up multiple playbin
objects concurrently (see #512382).
 2011-01-02 14:30:06 +00:00
Sebastian Dröge
459484bc4d Use GST_LICENSE, GST_PACKAGE_NAME and GST_PACKAGE_ORIGIN instead of hardcoding values where possible. Fixes bug #522212. 
Original commit message from CVS:
* ext/alsaspdif/alsaspdifsink.c:
* ext/gsm/gstgsm.c:
* ext/jack/gstjack.c:
* ext/libmms/gstmms.c:
* ext/neon/gstneonhttpsrc.c:
* ext/shout/gstshout.c:
* ext/timidity/gsttimidity.c:
* ext/timidity/gstwildmidi.c:
* gst/nuvdemux/gstnuvdemux.c:
* gst/tta/gsttta.c:
Use GST_LICENSE, GST_PACKAGE_NAME and GST_PACKAGE_ORIGIN instead
of hardcoding values where possible. Fixes bug #522212.
 2011-01-02 14:30:06 +00:00
Stefan Kost
be5f69be58 ext/jack/gstjackaudiosink.c: Add stdlib include here too. 
Original commit message from CVS:
* ext/jack/gstjackaudiosink.c: (gst_jack_ring_buffer_open_device),
(gst_jack_ring_buffer_acquire):
Add stdlib include here too.
 2011-01-02 14:30:06 +00:00
Stefan Kost
5b2d689774 ext/jack/gstjackaudiosink.c: Try t better name clients. properly handle return codes when re- establishing links. 
Original commit message from CVS:
* ext/jack/gstjackaudiosink.c: (gst_jack_ring_buffer_open_device),
(gst_jack_ring_buffer_acquire):
Try t better name clients. properly handle return codes when re-
establishing links.
 2011-01-02 14:30:06 +00:00
Paul Davis
a70ffeed7c ext/jack/gstjackaudioclient.c: Don't need to take the connection lock, it will not be used and could cause deadlocks. 
Original commit message from CVS:
Based on patch by: Paul Davis <paul at linuxaudiosystems dot com>
* ext/jack/gstjackaudioclient.c: (gst_jack_audio_unref_connection):
Don't need to take the connection lock, it will not be used and could
cause deadlocks.
 2011-01-02 14:30:06 +00:00
Paul Davis
60bcffa5ef ext/jack/: Make an object to manage client connections to the jack server which we will use in the future to run sele... 
Original commit message from CVS:
Includes patch by: Paul Davis <paul at linuxaudiosystems dot com>
* ext/jack/Makefile.am:
* ext/jack/gstjackaudioclient.c: (gst_jack_audio_client_init),
(jack_process_cb), (jack_sample_rate_cb), (jack_buffer_size_cb),
(jack_shutdown_cb), (connection_find),
(gst_jack_audio_make_connection), (gst_jack_audio_get_connection),
(gst_jack_audio_unref_connection),
(gst_jack_audio_connection_add_client),
(gst_jack_audio_connection_remove_client),
(gst_jack_audio_client_new), (gst_jack_audio_client_free),
(gst_jack_audio_client_get_client),
(gst_jack_audio_client_set_active):
* ext/jack/gstjackaudioclient.h:
Make an object to manage client connections to the jack server which we
will use in the future to run selected jack elements with the same jack
connection.
Make some stuff a bit more threadsafe.
Activate the jack client ASAP.
* ext/jack/gstjackaudiosink.c:
(gst_jack_audio_sink_allocate_channels),
(gst_jack_audio_sink_free_channels), (jack_process_cb),
(gst_jack_ring_buffer_open_device),
(gst_jack_ring_buffer_close_device),
(gst_jack_ring_buffer_acquire), (gst_jack_ring_buffer_release),
(gst_jack_audio_sink_class_init), (gst_jack_audio_sink_init),
(gst_jack_audio_sink_getcaps):
* ext/jack/gstjackaudiosink.h:
Use new client object to manage connections.
Don't remove and recreate all ports, try to reuse them.
 2011-01-02 14:30:06 +00:00
Wim Taymans
b0bfe6fcdd ext/jack/gstjackaudiosink.*: Improve docs. 
Original commit message from CVS:
* ext/jack/gstjackaudiosink.c: (jack_sample_rate_cb),
(jack_buffer_size_cb), (jack_shutdown_cb),
(gst_jack_ring_buffer_acquire):
* ext/jack/gstjackaudiosink.h:
Improve docs.
 2011-01-02 14:30:06 +00:00
Jan Schmidt
8beff1f58c ext/jack/.cvsignore: Ignore old files as requested by the build slave. 
Original commit message from CVS:
* ext/jack/.cvsignore:
Ignore old files as requested by the build slave.
 2011-01-02 14:30:06 +00:00
Wim Taymans
e2546f8275 ext/Makefile.am: Fix build. 
Original commit message from CVS:
* ext/Makefile.am:
Fix build.
* ext/jack/gstjackaudiosink.c: (jack_process_cb),
(jack_sample_rate_cb), (jack_buffer_size_cb), (jack_shutdown_cb),
(gst_jack_ring_buffer_acquire):
Small cleanups.
 2011-01-02 14:30:06 +00:00
Wim Taymans
32aaf1e95d Added fully functional jackaudiosink. 
Original commit message from CVS:
* configure.ac:
* ext/Makefile.am:
* ext/jack/Makefile.am:
* ext/jack/gstjack.c: (plugin_init):
* ext/jack/gstjack.h:
* ext/jack/gstjackaudiosink.c: (gst_jack_ring_buffer_get_type),
(gst_jack_ring_buffer_class_init), (jack_process_cb),
(jack_sample_rate_cb), (jack_buffer_size_cb), (jack_shutdown_cb),
(gst_jack_ring_buffer_init), (gst_jack_ring_buffer_dispose),
(gst_jack_ring_buffer_finalize),
(gst_jack_ring_buffer_open_device),
(gst_jack_ring_buffer_close_device),
(gst_jack_ring_buffer_acquire), (gst_jack_ring_buffer_release),
(gst_jack_ring_buffer_start), (gst_jack_ring_buffer_pause),
(gst_jack_ring_buffer_stop), (gst_jack_ring_buffer_delay),
(gst_jack_connect_get_type), (gst_jack_audio_sink_base_init),
(gst_jack_audio_sink_class_init), (gst_jack_audio_sink_init),
(gst_jack_audio_sink_set_property),
(gst_jack_audio_sink_get_property), (gst_jack_audio_sink_getcaps),
(gst_jack_audio_sink_create_ringbuffer):
* ext/jack/gstjackaudiosink.h:
Added fully functional jackaudiosink.
 2011-01-02 14:30:06 +00:00
Stefan Kost
8fa70e3622 Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent) 
Original commit message from CVS:
* ext/amrwb/gstamrwbdec.c: (gst_amrwbdec_class_init):
* ext/amrwb/gstamrwbenc.c: (gst_amrwbenc_class_init):
* ext/amrwb/gstamrwbparse.c: (gst_amrwbparse_class_init):
* ext/arts/gst_arts.c: (gst_arts_class_init):
* ext/artsd/gstartsdsink.c: (gst_artsdsink_class_init):
* ext/audiofile/gstafsink.c: (gst_afsink_class_init):
* ext/audiofile/gstafsrc.c: (gst_afsrc_class_init):
* ext/audioresample/gstaudioresample.c:
* ext/cdaudio/gstcdaudio.c: (gst_cdaudio_class_init):
* ext/directfb/dfbvideosink.c: (gst_dfbvideosink_class_init):
* ext/divx/gstdivxdec.c: (gst_divxdec_class_init):
* ext/hermes/gsthermescolorspace.c:
(gst_hermes_colorspace_class_init):
* ext/ivorbis/vorbisfile.c: (gst_ivorbisfile_class_init):
* ext/jack/gstjack.c: (gst_jack_class_init):
* ext/jack/gstjackbin.c: (gst_jack_bin_class_init):
* ext/lcs/gstcolorspace.c: (gst_colorspace_class_init):
* ext/libfame/gstlibfame.c: (gst_fameenc_class_init):
* ext/musicbrainz/gsttrm.c: (gst_musicbrainz_class_init):
* ext/nas/nassink.c: (gst_nassink_class_init):
* ext/shout/gstshout.c: (gst_icecastsend_class_init):
* ext/snapshot/gstsnapshot.c: (gst_snapshot_class_init):
* ext/sndfile/gstsf.c: (gst_sf_class_init):
* ext/swfdec/gstswfdec.c: (gst_swfdecbuffer_class_init),
(gst_swfdec_class_init):
* ext/tarkin/gsttarkindec.c: (gst_tarkindec_class_init):
* ext/tarkin/gsttarkinenc.c: (gst_tarkinenc_class_init):
* gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_class_init):
* gst/chart/gstchart.c: (gst_chart_class_init):
* gst/colorspace/gstcolorspace.c: (gst_colorspace_class_init):
* gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_class_init):
* gst/festival/gstfestival.c: (gst_festival_class_init):
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init):
* gst/filter/gstiir.c: (gst_iir_class_init):
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init):
* gst/librfb/gstrfbsrc.c: (gst_rfbsrc_class_init):
* gst/mixmatrix/mixmatrix.c: (gst_mixmatrix_class_init):
* gst/mpeg1sys/gstmpeg1systemencode.c:
(gst_system_encode_class_init):
* gst/mpeg1videoparse/gstmp1videoparse.c:
(gst_mp1videoparse_class_init):
* gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_class_init):
* gst/mpegaudioparse/gstmpegaudioparse.c:
(gst_mp3parse_class_init):
* gst/overlay/gstoverlay.c: (gst_overlay_class_init):
* gst/passthrough/gstpassthrough.c: (passthrough_class_init):
* gst/playondemand/gstplayondemand.c: (play_on_demand_class_init):
* gst/rtjpeg/gstrtjpegdec.c: (gst_rtjpegdec_class_init):
* gst/rtjpeg/gstrtjpegenc.c: (gst_rtjpegenc_class_init):
* gst/smooth/gstsmooth.c: (gst_smooth_class_init):
* gst/smoothwave/gstsmoothwave.c: (gst_smoothwave_class_init):
* gst/spectrum/gstspectrum.c: (gst_spectrum_class_init):
* gst/stereo/gststereo.c: (gst_stereo_class_init):
* gst/switch/gstswitch.c: (gst_switch_class_init):
* gst/tta/gstttadec.c: (gst_tta_dec_class_init):
* gst/tta/gstttaparse.c: (gst_tta_parse_class_init):
* gst/vbidec/gstvbidec.c: (gst_vbidec_class_init):
* gst/videocrop/gstvideocrop.c: (gst_video_crop_class_init):
* gst/virtualdub/gstxsharpen.c: (gst_xsharpen_class_init):
* gst/y4m/gsty4mencode.c: (gst_y4mencode_class_init):
* sys/cdrom/gstcdplayer.c: (cdplayer_class_init):
* sys/directsound/gstdirectsoundsink.c:
(gst_directsoundsink_class_init):
* sys/dxr3/dxr3audiosink.c: (dxr3audiosink_class_init):
* sys/dxr3/dxr3spusink.c: (dxr3spusink_class_init):
* sys/dxr3/dxr3videosink.c: (dxr3videosink_class_init):
* sys/qcam/gstqcamsrc.c: (gst_qcamsrc_class_init):
* sys/v4l2/gstv4l2colorbalance.c:
(gst_v4l2_color_balance_channel_class_init):
* sys/v4l2/gstv4l2tuner.c: (gst_v4l2_tuner_channel_class_init),
(gst_v4l2_tuner_norm_class_init):
* sys/ximagesrc/ximagesrc.c: (gst_ximagesrc_class_init):
Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
 2011-01-02 14:30:05 +00:00
Thomas Vander Stichele
cef77bfc9c rework build; add translations for v4l2 
Original commit message from CVS:
rework build; add translations for v4l2
 2011-01-02 14:30:05 +00:00
Stefan Kost
c038b32b44 renamed GST_FLAGS macros to GST_OBJECT_FLAGS moved bitshift from macro to enum definition 
Original commit message from CVS:
* examples/indexing/indexmpeg.c: (main):
* ext/artsd/gstartsdsink.c: (gst_artsdsink_open_audio),
(gst_artsdsink_close_audio), (gst_artsdsink_change_state):
* ext/artsd/gstartsdsink.h:
* ext/audiofile/gstafparse.c: (gst_afparse_open_file),
(gst_afparse_close_file):
* ext/audiofile/gstafparse.h:
* ext/audiofile/gstafsink.c: (gst_afsink_open_file),
(gst_afsink_close_file), (gst_afsink_chain),
(gst_afsink_change_state):
* ext/audiofile/gstafsink.h:
* ext/audiofile/gstafsrc.c: (gst_afsrc_open_file),
(gst_afsrc_close_file), (gst_afsrc_change_state):
* ext/audiofile/gstafsrc.h:
* ext/cdaudio/gstcdaudio.c: (gst_cdaudio_init):
* ext/directfb/directfbvideosink.c: (gst_directfbvideosink_init):
* ext/dts/gstdtsdec.c: (gst_dtsdec_init):
* ext/jack/gstjack.h:
* ext/jack/gstjackbin.c: (gst_jack_bin_init),
(gst_jack_bin_change_state):
* ext/musepack/gstmusepackdec.c: (gst_musepackdec_init):
* ext/musicbrainz/gsttrm.c: (gst_musicbrainz_init):
* ext/nas/nassink.c: (gst_nassink_open_audio),
(gst_nassink_close_audio), (gst_nassink_change_state):
* ext/nas/nassink.h:
* ext/polyp/polypsink.c: (gst_polypsink_init):
* ext/sdl/sdlvideosink.c: (gst_sdlvideosink_change_state):
* ext/sdl/sdlvideosink.h:
* ext/smoothwave/gstsmoothwave.c: (gst_smoothwave_init):
* ext/sndfile/gstsf.c: (gst_sf_set_property),
(gst_sf_change_state), (gst_sf_release_request_pad),
(gst_sf_open_file), (gst_sf_close_file), (gst_sf_loop):
* ext/sndfile/gstsf.h:
* ext/swfdec/gstswfdec.c: (gst_swfdec_init):
* ext/tarkin/gsttarkindec.c: (gst_tarkindec_init):
* gst/apetag/apedemux.c: (gst_ape_demux_init):
* gst/cdxaparse/gstcdxaparse.c: (gst_cdxaparse_init):
* gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_init):
* gst/festival/gstfestival.c: (gst_festival_change_state):
* gst/festival/gstfestival.h:
* gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_init):
* gst/multifilesink/gstmultifilesink.c: (gst_multifilesink_init),
(gst_multifilesink_set_location), (gst_multifilesink_open_file),
(gst_multifilesink_close_file), (gst_multifilesink_next_file),
(gst_multifilesink_pad_query), (gst_multifilesink_handle_event),
(gst_multifilesink_chain), (gst_multifilesink_change_state):
* gst/multifilesink/gstmultifilesink.h:
* gst/videodrop/gstvideodrop.c: (gst_videodrop_init):
* sys/cdrom/gstcdplayer.c: (cdplayer_init):
* sys/dxr3/dxr3audiosink.c: (dxr3audiosink_init),
(dxr3audiosink_open), (dxr3audiosink_close),
(dxr3audiosink_chain_pcm), (dxr3audiosink_chain_ac3),
(dxr3audiosink_change_state):
* sys/dxr3/dxr3audiosink.h:
* sys/dxr3/dxr3spusink.c: (dxr3spusink_init), (dxr3spusink_open),
(dxr3spusink_close), (dxr3spusink_chain),
(dxr3spusink_change_state):
* sys/dxr3/dxr3spusink.h:
* sys/dxr3/dxr3videosink.c: (dxr3videosink_init),
(dxr3videosink_open), (dxr3videosink_close),
(dxr3videosink_write_data), (dxr3videosink_change_state):
* sys/dxr3/dxr3videosink.h:
* sys/glsink/glimagesink.c: (gst_glimagesink_init):
* sys/qcam/gstqcamsrc.c: (gst_qcamsrc_change_state),
(gst_qcamsrc_open), (gst_qcamsrc_close):
* sys/qcam/gstqcamsrc.h:
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_init):
* sys/vcd/vcdsrc.c: (gst_vcdsrc_set_property), (gst_vcdsrc_get),
(gst_vcdsrc_open_file), (gst_vcdsrc_close_file),
(gst_vcdsrc_change_state), (gst_vcdsrc_recalculate):
* sys/vcd/vcdsrc.h:
renamed GST_FLAGS macros to GST_OBJECT_FLAGS
moved bitshift from macro to enum definition
 2011-01-02 14:30:05 +00:00
Jan Schmidt
473a76d221 Fix up all the state change functions. 
Original commit message from CVS:
Fix up all the state change functions.
 2011-01-02 14:30:05 +00:00
Benjamin Otte
567ac42d7e fixes for G_DISABLE_ASSERT and friends 
Original commit message from CVS:
* examples/dynparams/filter.c: (ui_control_create):
* examples/gstplay/player.c: (print_tag):
* ext/alsa/gstalsa.c: (gst_alsa_request_new_pad):
* ext/gdk_pixbuf/gstgdkanimation.c:
(gst_gdk_animation_iter_may_advance):
* ext/jack/gstjack.c: (gst_jack_request_new_pad):
* ext/mad/gstid3tag.c: (gst_mad_id3_to_tag_list),
(tag_list_to_id3_tag_foreach), (gst_id3_tag_handle_event):
* ext/vorbis/oggvorbisenc.c: (gst_oggvorbisenc_get_tag_value):
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_get_tag_value):
* ext/xine/xineaudiodec.c: (gst_xine_audio_dec_chain):
* gst-libs/gst/media-info/media-info-test.c: (print_tag):
* gst/sine/demo-dparams.c: (main):
* gst/tags/gstvorbistag.c: (gst_tag_to_vorbis_comments):
* testsuite/alsa/formats.c: (create_pipeline):
* testsuite/alsa/sinesrc.c: (sinesrc_force_caps), (sinesrc_get):
fixes for G_DISABLE_ASSERT and friends
* gst/typefind/gsttypefindfunctions.c: (aac_type_find),
(mp3_type_frame_length_from_header), (mp3_type_find),
(plugin_init):
require mp3 typefinding to have at least MIN_HEADERS valid headers
add typefinding for AAC adts files
 2011-01-02 14:30:04 +00:00
Stéphane Loeuillet
da195d3211 second batch : remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc (in ... 
Original commit message from CVS:
second batch :
remove ',' at end of enums as they could confuse older gcc, foreign compilers (forte) and gtk-doc
(in gst-plugins/ext/ this time)
 2011-01-02 14:30:04 +00:00
Thomas Vander Stichele
f88ab240e5 don't mix tabs and spaces 
Original commit message from CVS:
don't mix tabs and spaces
 2011-01-02 14:30:04 +00:00
Johan Dahlin
44f714a5fd *.h: Revert indenting 
Original commit message from CVS:
* *.h: Revert indenting
 2011-01-02 14:30:04 +00:00
Thomas Vander Stichele
c2bcfc8446 gst-indent 
Original commit message from CVS:
gst-indent
 2011-01-02 14:30:04 +00:00
David Schleef
d952e72d42 Remove all usage of gst_pad_get_caps(), and replace it with gst_pad_get_allowed_caps() or gst_pad_get_negotiated_cap(). 
Original commit message from CVS:
Remove all usage of gst_pad_get_caps(), and replace it with
gst_pad_get_allowed_caps() or gst_pad_get_negotiated_cap().
 2011-01-02 14:30:04 +00:00
David Schleef
9d2631e7d6 Merge CAPS branch 
Original commit message from CVS:
Merge CAPS branch
 2011-01-02 14:30:04 +00:00
Benjamin Otte
93cb77d0f0 removed GST_*_CAST. Disabling of type checking is done in glib. 
Original commit message from CVS:
removed GST_*_CAST. Disabling of type checking is done in glib.
 2011-01-02 14:30:04 +00:00
Andy Wingo
d8b76118bd remove copyright field from plugins 
Original commit message from CVS:
remove copyright field from plugins
 2011-01-02 14:30:03 +00:00
Ronald S. Bultje
56906e2c2e Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes fro... 
Original commit message from CVS:
Remove all config.h includes from header files, add it to each source file and remove duplicate config.h includes from several source files
 2011-01-02 14:30:03 +00:00
Iain Holmes
04c63f1f5c Jack fixed too 
Original commit message from CVS:
Jack fixed too
 2011-01-02 14:30:03 +00:00
David Schleef
bc5b62d938 change gst/bytestream.h to gst/bytestream/bytestream.h 
Original commit message from CVS:
change gst/bytestream.h to gst/bytestream/bytestream.h
 2011-01-02 14:30:03 +00:00
Benjamin Otte
bbc133ec8f merge TYPEFIND branch. Major changes: 
Original commit message from CVS:
merge TYPEFIND branch. Major changes:
- totally reworked type(find) system
- all typefind functions are in gst/typefind now
- more typefind functions then before
- some plugins might fail to compile now because I don't have them installed and they
a) require bytestream or
b) haven't had their typefind fixed.
Please fix those plugins and put the typefind functions into gst/typefind if they don't have dependencies
 2011-01-02 14:30:03 +00:00
Andy Wingo
45aedd595e /GstBuffer/GstData/ in the API where you can pass events. Fix the plugins to deal with that. Fixes #113488. 
Original commit message from CVS:
/GstBuffer/GstData/ in the API where you can pass events. Fix the plugins to deal with that. Fixes #113488.
 2011-01-02 14:30:03 +00:00
Ronald S. Bultje
fc1b8b479f New typefind system: bytestream is now part of the core all plugins have been modified to use this new typefind syste... 
Original commit message from CVS:
New typefind system:
* bytestream is now part of the core
* all plugins have been modified to use this new typefind system
* asf typefinding added
* mpeg video stream typefiding removed because it's broken
* duplicate typefind entries removed
* extra id3 typefinding added, because we've seen 4 types of files
(riff/wav, flac, vorbis, mp3) with id3 headers and each of these needs
to work. Instead, I've added an id3 element and let it redo typefiding
after the id3 header. this needs a hack because spider only typefinds
once. We can remove this hack once spider supports multiple typefinds.
* with all this, mp3 typefinding is semi-rewritten
* id3 typefinding in flac/vorbis is removed, it's no longer needed
* fixed spider and gst-typefind to use this, too.
* Other general cleanups
 2011-01-02 14:30:03 +00:00
Andy Wingo
b5802bdbb6 conform to the buffer-frames props entry -- much nicer now... 
Original commit message from CVS:
conform to the buffer-frames props entry -- much nicer now...
 2011-01-02 14:30:03 +00:00
David Schleef
b87729bd69 Remove redundant plugindir definition 
Original commit message from CVS:
Remove redundant plugindir definition
 2011-01-02 14:30:03 +00:00
Leif Johnson
25dd7cc50c + changes for new float caps without slope/intercept + some category changes for plugins 
Original commit message from CVS:
+ changes for new float caps without slope/intercept
+ some category changes for plugins
 2011-01-02 14:30:03 +00:00
Ronald S. Bultje
8bdd27fd62 New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as descri... 
Original commit message from CVS:
New mimetypes gone into effect today - this commit changes all old mimetypes over to the new mimetypes spec as described in the previous commit's document. Note: some plugins will break, some pipelines will break, expect HEAD to be broken or at least not 100% working for a few days, but don't forget to report bugs
 2011-01-02 14:30:03 +00:00
David Schleef
9a04390288 fix type punning 
Original commit message from CVS:
fix type punning
 2011-01-02 14:30:03 +00:00