Commit graph

12883 commits

Author SHA1 Message Date
Wim Taymans
4397c8ffbf rtpdepay: remove payload type restrictions
Remove the pt restrictions for all the depayloaders that have an
encoding-name. We can use this to autoplug decoders.
Remove the encoding-name for all the payloaders with a fixed payload
type.
We now either have an encoding-name or a pt in the sinkpad caps of
a depayloader.

See https://bugzilla.gnome.org/show_bug.cgi?id=639292
2013-01-28 12:41:04 +01:00
Marc Leeman
bab2f3c92b rtp: remove payload requirements from selected depayloaders
encoding name is required in the caps and is a better fit for autoplugging than
the pt value. Hardware manufacturers have a bad habit of skimming through RFCs
and in this case; use unassigned numbers for encoders instead of dynamic
numbers.

In essence, this patch will add support for a lot of Bosch hardware encoders
without breaking autoplugging.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=639292
2013-01-28 12:23:41 +01:00
B.Prathibha
7bb368ee4c tests: use g_timeout_add_seconds instead of g_timeout_add
https://bugzilla.gnome.org/show_bug.cgi?id=692615
2013-01-27 15:38:12 +00:00
Mark Nauwelaerts
a1a579afeb qtdemux: push mode: only parse moov 1 once
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=691570
2013-01-27 12:54:20 +01:00
Tim-Philipp Müller
47fccbe635 rtpdtmfsrc: fix compiler warning
gstrtpdtmfsrc.c: In function 'gst_dtmf_src_prepare_message.isra.1':
gstrtpdtmfsrc.c:669:3: error: 's' may be used uninitialized in this function
2013-01-26 22:58:29 +00:00
Olivier Crête
db5c3f4048 rtpdtmfdepay: Fix missing work in doc 2013-01-25 21:06:05 -05:00
Olivier Crête
6105510a7a tests: Add test for rtpdtmfdepay and rtpdtmfsrc 2013-01-25 21:05:39 -05:00
Olivier Crête
92f9a9d9ff rtpdtmfsrc: Post the messages after the clock wait
This way, the messages will be closer in time to when the packets are sent out
2013-01-25 20:45:43 -05:00
Olivier Crête
0d316b4f43 rtpdtmfsrc: Only set the duration when starting to send
The duration depends on the clock rate, which could change due to renegotiation
2013-01-25 20:45:43 -05:00
Olivier Crête
90497aa3cd rtpdtmfsrc: remove "ssrc" from caps
ssrc is uint and we don't have a uint range type
2013-01-25 20:45:43 -05:00
Tim-Philipp Müller
d62019fff2 qtmux: set language to 'undefined' instead of English by default 2013-01-24 21:08:51 +00:00
Olivier Crête
a31649e357 ximagesrc: Set the pixel aspect ratio correctly in the caps 2013-01-23 21:35:25 -05:00
Sjoerd Simons
00eed11d6a v4l2: Re-enable prepare-format emission
With the port to gstreamer 1.0 the prepare-format signal stopped being
emitted. Start emitting this again for use in uvch264src.  While there
change the emission to include the caps for extra flexibility instead of
fource, width, height.

https://bugzilla.gnome.org/show_bug.cgi?id=692042
2013-01-23 21:06:16 -05:00
Benjamin Gaignard
3d1496285c autogen.sh: allow calling from out-of-tree
Signed-off-by: Benjamin Gaignard <benjamin.gaignard@st.com>

https://bugzilla.gnome.org/show_bug.cgi?id=692309
2013-01-23 10:28:13 +00:00
Mark Nauwelaerts
0777a600e3 audioparsers: sbc: fix bogus compiler warning
gst-plugins-good/gst/audioparsers/gstsbcparse.c: In function 'gst_sbc_parse_handle_frame':
gst-plugins-good/gst/audioparsers/gstsbcparse.c:210:32: error: 'ch_mode' may be used uninitialized i
2013-01-22 19:26:09 +01:00
Tim-Philipp Müller
0c9b039c22 pulsesink: don't error out if pa_stream_proplist_update() with new tags fails
Shouldn't really happen these days, but if it does, it's not really
a problem either.

https://bugzilla.gnome.org/show_bug.cgi?id=656068
2013-01-19 13:27:48 +00:00
Tim-Philipp Müller
066600c18b tests: skip souphttpsrc tests if there is no local http server to use
Skip tests if the server couldn't be started or we can't connect
to it for some reason (e.g. draconic build bot environments).
2013-01-16 18:03:44 +00:00
Thijs Vermeir
16128f0234 autoparsers: use appropriate printf format for gsize 2013-01-16 14:32:56 +01:00
Martin Pitt
b46dcf379a tests: use _1_0 variants for the various registry variables
These override the variants without version suffix. Makes 'make check' work
properly in environments that set the suffixed variant for 1.0, such as
jhbuild.
2013-01-16 11:10:19 +00:00
Alexey Chernov
a8fe984d65 osxvideosink: Fix crash in osxvideosink with external window output 2013-01-16 11:43:56 +01:00
Alexey Chernov
77fde4b8ba osxvideosink: Make GstGLView propagate input events to its parent view
Fixes bug #691832
2013-01-16 11:38:16 +01:00
Tim-Philipp Müller
9455a3aee1 rtpsbcpay: update some fields in the caps to their new name
and to match the parser. "mode" got renamed to "channel-mode"
and "allocation" to "allocation-method".
2013-01-16 10:19:36 +00:00
Tim-Philipp Müller
f307c6d491 docs: add sbcparse and rtpsbcpay to plugin docs 2013-01-15 17:45:30 +00:00
Tim-Philipp Müller
9f7a949773 audioparsers: add SBC audio parser
From-scratch rewrite, the bluez one was useless and broken.

https://bugzilla.gnome.org/show_bug.cgi?id=690582
2013-01-15 17:45:30 +00:00
Tim-Philipp Müller
b6de647901 Automatic update of common submodule
From a72faea to a942293
2013-01-15 15:05:04 +00:00
Tim-Philipp Müller
39ef892938 rtp: import rtpsbcpay from bluez and port to 1.0
Compiles, but not tested yet (sbc elements still need to be ported).

https://bugzilla.gnome.org/show_bug.cgi?id=690582
2013-01-10 12:43:50 +00:00
Olivier Crête
c6dea5d09c dtmf/spandsp: Move dtmfdetect to use libspandsp
Remove our copy of the tone_detect.c file and use the original
from libspandsp. Also move the element to the spandsp plugin.
2013-01-09 20:05:16 -05:00
Marcel Holtmann
4196feb659 rtpsbcpay: Remove workaround for compiler warnings 2013-01-10 00:18:03 +00:00
Marcel Holtmann
fe79c60d74 rtpsbcpay: Add pragma based workaround for GStreamer warnings 2013-01-10 00:18:03 +00:00
Marcel Holtmann
08e95e7249 rtpsbcpay: Update copyright information 2013-01-10 00:15:36 +00:00
Marcel Holtmann
7fa03c0076 rtpsbcpay: Fix signed/unsigned comparison issue within GStreamer plugin 2013-01-10 00:15:35 +00:00
Marcel Holtmann
27a6b0abfe rtpsbcpay: Update copyright information 2013-01-10 00:15:35 +00:00
Marcel Holtmann
f890079aae rtpsbcpay: First attempt in fixing compiler warnings (still needs cleanup) 2013-01-10 00:15:35 +00:00
Johan Hedberg
7d4f846112 rtpsbcpay: More coding style fixes 2013-01-10 00:15:35 +00:00
Luiz Augusto von Dentz
151ad9b28d rtpsbcpay: Remove possible extra memcpy for gstreamer plugin. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
69c8374b7c rtpsbcpay: Fix bug sending empty packages and remove a buffer copy. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
7b3e4356ea rtpsbcpay: Fix runtime warnings of gstreamer plugin. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
f74f061f3b rtpsbcpay: Update gstreamer plugin to use new sbc API. 2013-01-10 00:13:14 +00:00
Marcel Holtmann
b9be04f07b rtpsbcpay: Update copyright information 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
687400ecf4 rtpsbcpay: Fixes gstreamer caps and code cleanup. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
a4f9624261 rtpsbcpay: Fix gtreamer payloader sending fragmented frames. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
41e2f4f544 rtpsbcpay: Fix use of gstreamer plugin with rhythmbox and banshee and rtp timestamps. 2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz
96971cd323 rtpsbcpay: Make a2dpsink to act like a bin and split the payloader. 2013-01-10 00:13:14 +00:00
Wim Taymans
72402cc649 rtp: small improvements 2013-01-08 16:27:42 +01:00
Wim Taymans
af055d9574 jitterbuffer: refactor handle sync code
Move the code that combines the last SR packet and the current jitterbuffer sync
values into a sync structure, into its own function. We want to reuse this bit
later.
2013-01-07 15:50:33 +01:00
Wim Taymans
87f7d6b9bf rtp: include downstream latency in SR calculations
When we make a mapping between an RTP timestamp and an NTP timestamp, include
the downstream latency applied to the sinks. This makes it possible to have
both sinks run with different latencies and still have correct sync on the
client. It also is more correct because the RTP timestamp in the SR report will
actually correspond more closely to the NTP time it was sent on the server.
For pipelines with high latency on the sender side, this actually allows a
GStreamer receiver to perform synchronisation instead of dropping the RTCP
packets.
2013-01-07 15:45:10 +01:00
Wim Taymans
c631ed3300 rtpsession: don't cast event functions
There is no need to cast the event functions and only causes problems later when
we change the signature later and things silently compiles wrong code.
2013-01-07 14:25:14 +01:00
Wim Taymans
8dcde8b3ea rtp: more debug 2013-01-07 14:23:34 +01:00
Wim Taymans
6b7d05ac57 rtpsession: improve debug 2013-01-07 14:22:48 +01:00
Tim-Philipp Müller
cf1f6aff0d udpsrc: sanity check size of available packet data for reading to avoid memory waste
On Windows and OS/X, _get_available_bytes() may not return the size
of the next pending packet, but the size of all pending packets in
the kernel-side buffer, which might be rather large depending on
configuration. Sanity-check the size returned by _get_available_bytes()
to make sure we never allocate more memory than the max. size for
a packet, if it's an IPv4 socket.

https://bugzilla.gnome.org/show_bug.cgi?id=610364
2013-01-04 14:00:55 +00:00