Commit graph

306 commits

Author SHA1 Message Date
Wim Taymans
ecaea36c3d update for memory api changes 2012-03-15 13:36:17 +01:00
Wim Taymans
89105970f0 flacenc: fix streamheaders
Fix the caps of flacenc, the reference encoder only support 24 bits in
32 bits.
Set streamheader on output caps.
2012-03-13 12:40:37 +01:00
Wim Taymans
a51ce46d90 flacenc: fix event handling
Fix dodgy segment event handling
Chain up to parent event handler
2012-02-27 13:05:33 +01:00
Tim-Philipp Müller
61d3a215a0 Merge commit '38516ad367128d83f9e156529018adb4433cd328' into 0.11
Conflicts:
	ext/pulse/pulseaudiosink.c
	gst/audioparsers/gstmpegaudioparse.c
2012-02-27 00:48:57 +00:00
Tim-Philipp Müller
3e9f191262 flacenc: fix get_caps function some more so that all structures have channel info
Set channels and channel-layout on the right structure; that is, the
structure we are going to append to the caps we are building, and not
the structure we are using as a template for all the structures. Fixes
first structure of the returned caps not having any channel info set
on it.
2012-02-22 17:39:16 +00:00
Tim-Philipp Müller
f0b076212f flacenc: microoptimisation: avoid unnecessary list and string copies 2012-02-22 17:09:25 +00:00
Tim-Philipp Müller
9ce663f04d flacenc: audio caps have a *list* of formats, not an array of formats
A list of things in caps is something where one is picked in the
course of negotiation. An array is always something that only makes
sense as a whole in that order.
2012-02-22 17:03:42 +00:00
Mark Nauwelaerts
38516ad367 flacenc: remove post-port bogus _unref 2012-02-22 18:03:11 +01:00
Tim-Philipp Müller
b7e96ebe37 flacenc: remove bogus pad locking that causes deadlocks
It's not clear why the pad object lock is taken here. But
gst_pad_{has,get}_current_caps() will try to take the lock
as well and deadlock, since it's not recursive.
2012-02-22 17:00:19 +00:00
Tim-Philipp Müller
9c5c33790a flacenc: set right number of channels on caps in get_caps function 2012-02-22 16:59:42 +00:00
Wim Taymans
225e98d623 Merge branch 'master' into 0.11
Conflicts:
	ext/flac/gstflacenc.c
	ext/jack/gstjackaudioclient.c
	ext/jack/gstjackaudiosink.c
	ext/jack/gstjackaudiosrc.c
	ext/pulse/plugin.c
	ext/shout2/gstshout2.c
	gst/matroska/matroska-mux.c
	gst/rtp/gstrtph264pay.c
2012-02-10 16:23:14 +01:00
Mark Nauwelaerts
abc8c162ed flacdec: shift in proper direction for audio sample conversion 2012-02-09 22:09:31 +01:00
Vincent Penquerc'h
5ff31d446e flacenc: fix event leak when there is no peer on the src pad 2012-02-03 14:53:31 +00:00
Sebastian Dröge
161229a384 flac: Use new audio encoder/decoder base class API for srcpad caps 2012-02-01 16:27:47 +01:00
Wim Taymans
bb2bd604e0 update for HEADER flag 2012-01-30 17:16:51 +01:00
Wim Taymans
b4630dd3e0 more memory API porting 2012-01-25 12:30:29 +01:00
Mark Nauwelaerts
1911812572 flacdec: improve upstream peer duration querying
... to avoid accepting unhandled duration query result.
2012-01-20 17:10:19 +01:00
Wim Taymans
1584806634 port to new gthread API 2012-01-19 11:33:53 +01:00
Tim-Philipp Müller
8580dd86c9 eqMerge remote-tracking branch 'origin/master' into 0.11
Conflicts:
	ext/jack/gstjackaudiosink.c
	ext/jack/gstjackaudiosrc.c
	gst/matroska/matroska-mux.c
	gst/matroska/matroska-read-common.c
	gst/rtpmanager/gstrtpssrcdemux.c
2012-01-12 23:48:50 +00:00
Vincent Penquerc'h
483514528a flacenc: do not drop the first data buffer on the floor (and leak it either) 2012-01-12 10:30:56 +00:00
Sebastian Dröge
93e3ed5a86 Merge branch 'master' into 0.11
Conflicts:
	ext/cairo/gsttextoverlay.c
	ext/pulse/pulseaudiosink.c
	gst/audioparsers/gstaacparse.c
	gst/avi/gstavimux.c
	gst/flv/gstflvmux.c
	gst/interleave/interleave.c
	gst/isomp4/gstqtmux.c
	gst/matroska/matroska-demux.c
	gst/matroska/matroska-mux.c
	gst/matroska/matroska-mux.h
	gst/matroska/matroska-read-common.c
	gst/multifile/gstmultifilesink.c
	gst/multipart/multipartmux.c
	gst/shapewipe/gstshapewipe.c
	gst/smpte/gstsmpte.c
	gst/udp/gstmultiudpsink.c
	gst/videobox/gstvideobox.c
	gst/videocrop/gstaspectratiocrop.c
	gst/videomixer/videomixer.c
	gst/videomixer/videomixer2.c
	gst/wavparse/gstwavparse.c
	po/ja.po
	po/lv.po
	po/sr.po
	tests/check/Makefile.am
	tests/check/elements/qtmux.c
	tests/check/elements/rgvolume.c
2012-01-10 14:32:32 +01:00
Sebastian Dröge
a22a566c0b flac: Port to the new raw audio caps 2012-01-06 09:40:55 +01:00
Tim-Philipp Müller
b8b8454bcb Suppress deprecation warnings in selected files, for g_static_rec_mutex_* mostly
GStaticRecMutex is part of our API/ABI, not much we can do here
in 0.10 for most of these.
2011-12-12 09:46:27 +00:00
Vincent Penquerc'h
c0e101e93f various: fix pad template leaks
https://bugzilla.gnome.org/show_bug.cgi?id=662664
2011-11-28 13:30:27 +00:00
Tim-Philipp Müller
736a484129 More printf format warning fixes 2011-11-22 01:40:39 +00:00
Wim Taymans
07cc855b24 Merge branch 'master' into 0.11
Conflicts:
	ext/speex/gstspeexenc.c
	gst/rtpmanager/rtpsession.c
2011-11-17 17:17:11 +01:00
Wim Taymans
105650127e add parent to pad functions 2011-11-17 15:02:55 +01:00
Mark Nauwelaerts
413f445455 flacenc: reset tag setter interface when appropriate 2011-11-16 19:06:07 +01:00
Wim Taymans
e7918a5aba _query_peer_*() -> _peer_query_*() 2011-11-15 18:04:44 +01:00
Vincent Penquerc'h
8548b2c777 flacdec: fix spurious timestamp discontinuity
We need to tell the base class that we're dropping buffers,
so it drops the input timestamps corresponding to these.
Otherwise, the first actual audio buffers we output will be
stamped with those - GST_CLOCK_TIMESTAMP_NONE. That mismatch
between input buffer count and output buffer count will stay
while playing. With enough headers and long enough buffer
durations, the sink will have played enough before receiving
the first valid timestamp (usually 0), and will trigger an
audible discontinuity.
2011-11-15 13:36:15 +00:00
Wim Taymans
7e12b58e37 update for adapter api changes 2011-11-10 18:32:58 +01:00
Vincent Penquerc'h
5a73374f2c flacdec: fix off by one between granpos and last_stop 2011-11-07 12:38:10 +00:00
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
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
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
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
e9df54819c Merge branch 'master' into 0.11 2011-08-24 14:16:44 +02:00
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
ce1e7cb108 Merge branch 'master' into 0.11
Conflicts:
	ext/flac/gstflacdec.c
2011-08-17 15:52:18 +02:00