Commit graph

196 commits

Author SHA1 Message Date
Tim-Philipp Müller
5ab46eff83 pulsesink: fix caps leak in acceptcaps function 2012-10-20 11:32:27 +01:00
Tim-Philipp Müller
d2fdc26c38 pulsesink: in accept_caps() check if ring buffer is NULL before de-referencing
And sprinkle some thread-safety (take object lock for
accessing ring buffer, and pa main loop lock for the
context).

https://bugzilla.gnome.org/show_bug.cgi?id=683782
2012-10-19 15:11:07 +01:00
Arun Raghavan
f46475ee37 pulsesink: Specify endianness in IEC 61937 payloading
Corresponds to an API change in gst-plugins-base.

https://bugzilla.gnome.org/show_bug.cgi?id=678021
2012-09-19 09:18:19 +05:30
Wim Taymans
148ab7539b pulse: improve debug 2012-09-06 10:43:52 +02:00
Tim-Philipp Müller
4bb52bbadf docs: gst-launch -> gst-launch-1.0 and ffmpegcolorspace -> videoconvert 2012-08-27 21:20:30 +01:00
Wim Taymans
373333c2b3 pulsesink: improve debug 2012-04-25 10:29:45 +02:00
Wim Taymans
c0140982ee pulsesink: start unmuted when requested
When we explicitely set the mute property to FALSE, connect to pulseaudio with
the PA_STREAM_START_UNMUTED flag set, otherwise pulseaudio will use its
previously used value (which might start the stream muted).

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=672401
2012-04-25 10:29:45 +02:00
Sebastian Dröge
d99eb6d2cb Update everything for the removal of the interface library and mixer/tuner interfaces 2012-04-13 13:15:11 +02:00
Tim-Philipp Müller
e09ae5736d Use new gst_element_class_set_static_metadata() 2012-04-10 00:51:41 +01:00
Sebastian Dröge
0b517ce9fb Merge branch '0.11' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-good into 0.11 2012-01-25 12:49:34 +01:00
Sebastian Dröge
10554b271f Merge branch 'master' into 0.11
Conflicts:
	ext/flac/gstflacdec.c
	ext/jpeg/gstjpegenc.c
	ext/pulse/pulsesink.c
	sys/v4l2/gstv4l2src.c
2012-01-25 12:49:11 +01:00
Wim Taymans
b4630dd3e0 more memory API porting 2012-01-25 12:30:29 +01:00
Mark Nauwelaerts
3168b77e04 pulsesink: additional error condition checking 2012-01-20 17:10:14 +01:00
Wim Taymans
1584806634 port to new gthread API 2012-01-19 11:33:53 +01:00
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
Tim-Philipp Müller
ff74718616 pulse: remove pulseaudiosink helper bin
This is causing us lots of headaches in 0.10 and needs to be done
differently and properly in 0.11. playbin or decodebin should
reconfigure themselves based on reconfigure events, for example.
2011-12-25 22:21:36 +00:00
Tim-Philipp Müller
2799bcd32e pulse: update for ring buffer audio format type enum rename 2011-12-25 21:45:45 +00:00
Wim Taymans
4b8975f867 update for removed property probe 2011-12-21 11:59:46 +01:00
Tim-Philipp Müller
66f6e12888 Work around deprecated thread API in glib master
Add private replacements for deprecated functions such as
g_mutex_new(), g_mutex_free(), g_cond_new() etc., mostly
to avoid the deprecation warnings. We'll change these
over to the new API once we depend on glib >= 2.32.
2011-12-12 09:46:27 +00:00
Tim-Philipp Müller
adb15bf34a pulse: rename "client" properties to "client-name"
Better name, but also matches the property on the jack
elements (where "client" is used for something else).
2011-12-09 16:04:56 +00:00
Wim Taymans
5bfc7b4bfe update for moved audio interfaces 2011-11-30 07:57:40 +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
eeaa9e0bbc pulseaudio: require pulseaudio >= 1.0 2011-11-26 13:54:22 +00:00
Wim Taymans
b2d508ac40 update for _get_caps() -> _query_caps() 2011-11-15 18:04:44 +01:00
Wim Taymans
b0ccc61ed3 Merge branch 'master' into 0.11
Conflicts:
	ext/pulse/pulseaudiosink.c
2011-11-11 19:24:27 +01:00
Mark Nauwelaerts
37c8abcdbd pulsesink: do not leak clientname when setting up property 2011-11-11 14:59:04 +01:00
Wim Taymans
3d9d2c6c05 update for audiobase* rename 2011-11-11 12:01:17 +01:00
Wim Taymans
86e33bc46b audio: update for base class rename 2011-11-11 11:53:45 +01:00
Wim Taymans
1ad11e307a update for ringbuffer change 2011-11-11 11:24:00 +01:00
René Stadler
3293b88ea1 pulsesink: fix compilation with pulseaudio 0.9 2011-11-10 21:37:38 +01:00
Wim Taymans
00d3f3a454 fix for audio clock change 2011-11-10 13:50:34 +01:00
Wim Taymans
aa0b2b7ea7 updates for new acceptcaps query 2011-11-09 17:38:03 +01:00
Wim Taymans
9a8a8e72c8 structure: fix for api update 2011-11-02 09:06:37 +01:00
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
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
Thiago Santos
358767e217 pulsesink: Fixing getcaps function
Update getcaps function to 0.11 API
2011-10-09 21:19:27 -03:00
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
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
Wim Taymans
e204c5934c -good: port to new audio caps 2011-09-06 13:16:27 +02:00
Arun Raghavan
bd604175c5 pulsesink: Trivial indentation fix 2011-08-23 22:48:34 +05:30
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
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
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
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