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15980 commits 44 branches 1507 tags 445 MiB
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174 commits

Author SHA1 Message Date
Vineeth TM
1071309870 good: use new gst_element_class_add_static_pad_template() 
https://bugzilla.gnome.org/show_bug.cgi?id=763076
 2016-03-24 14:32:20 +02:00
Thiago Santos
a1aa942acf audioencoders: use template subset check for accept-caps 
It is faster than doing a query that propagates downstream and
should be enough

Elements: speexenc, wavpackenc, mulawenc, alawenc
 2015-08-16 14:30:57 -03:00
Thiago Santos
65676c22ee audiodecoders: use default pad accept-caps handling 
Avoids useless check of downstream caps when handling an
accept-caps query

Elements: flacdec, speexdec, wavpackdec, mulawdec, alawdec
 2015-08-15 11:46:34 -03:00
Tim-Philipp Müller
c53747bdf5 speex: remove support for ancient speex versions 2014-11-22 21:28:35 +00:00
Vincent Penquerc'h
ca9528d0b0 speexenc: update output segment stop time to match clipped samples 
This will let oggmux generate a granpos on the last page that properly
represents the clipped samples at the end of the stream.
 2014-10-30 14:43:22 +00:00
Ananda
ec3af50cc2 speex: Fix segfault when resetting the codecs multiple times 
https://bugzilla.gnome.org/show_bug.cgi?id=738793
 2014-10-23 10:30:26 +02:00
Sebastian Rasmussen
485da06b14 speexenc: Improve annotation of internal function 
https://bugzilla.gnome.org/show_bug.cgi?id=734542
 2014-08-10 11:17:23 +01:00
Edward Hervey
9843b08e53 speexenc: add missing va_end in variadic function 
Coverity 1139944
 2014-06-09 10:39:20 +02:00
Vincent Penquerc'h
74c93b8fc7 speexdec: remove dead code 
fpp can never equal 0 here, or the loop would not execute at all.
Zero fpp was possible before as the loop condition was allowing
it specifically, but no more.

Coverity 1139681
 2014-05-02 09:45:07 +01:00
Tim-Philipp Müller
c9597298f9 docs: remove outdated and pointless 'Last reviewed' lines from docs 
They are very confusing for people, and more often than not
also just not very accurate. Seeing 'last reviewed: 2005' in
your docs is not very confidence-inspiring. Let's just remove
those comments.
 2014-04-26 23:35:17 +01:00
Sebastian Dröge
c880e36779 speexdec: Require caps to be set before accepting any data 2013-12-05 12:13:33 +01:00
Sebastian Dröge
b0b0557c48 gst: Add better support for static plugins 2013-04-15 15:54:11 +02:00
Debarshi Ray
8c44361bca speexdec: Don't unmap or finish_frame an invalid GstBuffer 
https://bugzilla.gnome.org/show_bug.cgi?id=687464
 2012-11-06 19:49:50 +00:00
Tim-Philipp Müller
230cf41cc9 Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
 2012-11-04 00:07:18 +00:00
Tim-Philipp Müller
4bb52bbadf docs: gst-launch -> gst-launch-1.0 and ffmpegcolorspace -> videoconvert 2012-08-27 21:20:30 +01:00
Tim-Philipp Müller
c074bfd0b9 gst_tag_list_free -> gst_tag_list_unref 2012-08-04 16:10:16 +01:00
Tim-Philipp Müller
e09ae5736d Use new gst_element_class_set_static_metadata() 2012-04-10 00:51:41 +01:00
Sebastian Dröge
aa2cd462da gst: Update for GST_PLUGIN_DEFINE() API changes 2012-04-05 17:36:38 +02:00
Sebastian Dröge
5cdd49bf25 gst: Update versioning 2012-04-04 14:37:47 +02:00
Sebastian Dröge
df946f603f speexenc: Use new gst_audio_encoder_set_headers() API 2012-03-30 12:53:44 +02:00
Sebastian Dröge
b16f5637e8 ext: Update for GstAudioEncoder API changes 2012-03-30 12:18:45 +02:00
Wim Taymans
c44cd8f55b Merge branch 'master' into 0.11 
unport gdkpixbuf
not merged: https://bugzilla.gnome.org/show_bug.cgi?id=654850

Conflicts:
	docs/plugins/Makefile.am
	docs/plugins/gst-plugins-good-plugins-docs.sgml
	docs/plugins/gst-plugins-good-plugins-sections.txt
	docs/plugins/gst-plugins-good-plugins.hierarchy
	docs/plugins/inspect/plugin-avi.xml
	docs/plugins/inspect/plugin-png.xml
	ext/flac/gstflacdec.c
	ext/flac/gstflacdec.h
	ext/libpng/gstpngdec.c
	ext/libpng/gstpngenc.c
	ext/speex/gstspeexdec.c
	gst/audioparsers/gstflacparse.c
	gst/flv/gstflvmux.c
	gst/rtp/gstrtpdvdepay.c
	gst/rtp/gstrtph264depay.c
 2012-03-22 11:53:24 +01:00
Wim Taymans
ecaea36c3d update for memory api changes 2012-03-15 13:36:17 +01:00
Mark Nauwelaerts
a199ad9001 speexdec: use base class tag handling helper 
... so as to ensure these to be handled and sent at proper time.
 2012-03-06 16:23:48 +01:00
Wim Taymans
36e6b25e73 speexenc: chain up to parent event handler 2012-02-27 13:09:31 +01:00
Sebastian Dröge
a67bd41d75 speex: 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
Sebastian Dröge
4b6a410be0 speex: Update for the new raw audio caps 2012-01-05 10:36:49 +01:00
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
Mark Nauwelaerts
7df8122322 speexenc: ensure to free allocated padded data 2011-11-16 19:08:05 +01:00
Mark Nauwelaerts
c0d86fd26f speexenc: reset tag setter interface when appropriate 2011-11-16 19:06:09 +01:00
Wim Taymans
e038ab5a0b tags: update for tag API removal 2011-11-02 12:09:20 +01:00
Tim-Philipp Müller
9cd17092d8 ext, gst: update for taglist API changes 2011-10-30 11:44:53 +00:00
Wim Taymans
aea9b5e8c8 Merge branch 'master' into 0.11 
Conflicts:
	ext/speex/gstspeexenc.c
 2011-10-10 11:48:20 +02:00
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
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
Vincent Penquerc'h
7e4574e968 speexenc: do not use invalid buffer timestamps 2011-09-19 09:37:58 +02:00
Wim Taymans
77ad0a1363 port more elements to new audio caps and API 2011-08-19 14:01:45 +02:00
Tim-Philipp Müller
ab62599832 speex: update for position/query/convert API changes 2011-07-28 11:38:31 +01:00
Wim Taymans
fdf5a49422 speex: port speex elements 2011-07-06 15:57:23 +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
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