Commit graph

1821 commits

Author SHA1 Message Date
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