Commit graph

179 commits

Author SHA1 Message Date
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
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
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
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
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
d59a00aa1c Merge branch 'master' into 0.11
Conflicts:
	ext/pulse/pulsesink.c
2011-07-04 11:48:13 +02:00
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
Wim Taymans
dea50d0b98 use G_DEFINE_TYPE 2011-04-19 17:07:18 +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
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
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
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
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
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