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Author SHA1 Message Date
Edward Hervey
b28c6ca0fb matroskademux: Remove useless variable. 
iret was never read outside of that loop, and is always being exited if
iret was != GST_FLOW_OK anyway.
 2009-04-18 18:51:28 +02:00
Edward Hervey
1086c63827 avidemux: Move 'res' to where it's actually being used. 
res was never used outside of that block except for a dead assignment.
 2009-04-18 18:51:28 +02:00
Edward Hervey
a299e86cfc audiofx: Remove unused variable. 
rz is never used in these methods.
 2009-04-18 18:51:28 +02:00
Edward Hervey
bd4f8576fe osxringbuffer: Run gst-indent. 2009-04-18 18:51:28 +02:00
Edward Hervey
14f715f978 ximage: Remove dead assignments. 
Those variables are not read after that point.
 2009-04-18 18:51:28 +02:00
Edward Hervey
0cb5b42d54 Remove trivial unused variables detected by CLang static analyzer. 2009-04-18 18:51:28 +02:00
Edward Hervey
cdb03bdc2b Remove blank {set|get}_property/change_state/finalize methods. 2009-04-18 18:51:27 +02:00
Edward Hervey
4a9e80720a Remove unused variables in _class_init 
Detected by LLVM's CLang static analyzer
 2009-04-18 18:51:27 +02:00
Jan Schmidt
06a4b80918 check: Check whether threads are already initialised before g_thread_init() 2009-04-18 14:05:16 +01:00
Josep Torra
dfb375daa1 rtspsrc: mark discont on the streams as was said the debug line 
After a seek mark all streams with discont as it was said in the debug line.
Fixes that buffers after a seek are generated without a valid timestamp.
 2009-04-18 14:32:40 +02:00
Josep Torra
ec2d6053a0 rtspsrc: map GST_RTSP_EEOF to EOS on server requests 
Permit properly handle the EOS condition when server report it in a request.
 2009-04-18 08:50:46 +02:00
Edward Hervey
f9d7640bc9 rtptheoradepay: Fix build on macosx. 
Use G_GSIZE_FORMAT instead of u.
 2009-04-18 08:39:57 +02:00
Wim Taymans
c052906590 pulsesink: fix sample offset calculation again 2009-04-16 22:51:54 +02:00
Tim-Philipp Müller
cdeb8ebb13 sunaudio: fix broken indentation of variable declarations 2009-04-15 19:33:16 +01:00
James Andrewartha
4a74e341ec sunaudio: remove some unused variables and goto labels 
Fixes #579070.
 2009-04-15 19:30:11 +01:00
James Andrewartha
ac48c2d211 rtph263pay: fix compilation on big-endian 
Some semicolons were missing from the big-endian structs in gstrtph263pay.h.
A GST_DEBUG call was missing a format specifier.

Fixes #579069
 2009-04-15 19:26:22 +02:00
Marco Ballesio
94d5d24cf0 qtdemux: implement 3GPP (TS 26.244 V8.0.0) Asset metadata handling, Fixes #132193 
Implements 3gpp iso metadata tags which are different from mov udta atoms.
 2009-04-15 20:14:19 +03:00
Peter Kjellerstedt
af7f3a50dd debugutils: Use G_BEGIN_DECLS/G_END_DECLS. 
Use G_BEGIN_DECLS/G_END_DECLS to avoid gst-indent messing up the
indentation due to extern "C" { }.
 2009-04-15 15:51:24 +02:00
Stefan Kost
7be792fa13 debug: rename debug to debugutils to avoid clash with --disable-debug. Fixes #562168 2009-04-15 16:13:34 +03:00
Stefan Kost
99fcc86ee4 debug: indent before renaming 2009-04-15 16:13:34 +03:00
Wim Taymans
787124dad6 g726depay: add property for aal2 force 2009-04-15 14:07:57 +02:00
Wim Taymans
0802ba8730 g726depay: implement RFC3551 packing 
We implemented the AAL2 packing, add the encoding-name for those to the caps and
a property to force AAL2 decoding (always TRUE for now).
Implement RFC3551 unpacking for regular G726.
See #567140.
 2009-04-15 13:56:17 +02:00
Wim Taymans
c34d5aa016 rtph263pay: fix build 2009-04-15 00:22:43 +02:00
Youness Alaoui
17d9cb3319 h263pay: various fixes 
Re-enable mode A support and a property to control it.
Fix memory leak of GstRtpH263PayBoundry objects.
Fix marker.
Fixes #509311
 2009-04-14 18:52:48 +02:00
Janin Kolenc
de2c489526 h263pay: Fix the payloader 
Fix the H263 payloader to be more RFC 2190 compliant.
See #509311
 2009-04-14 18:44:51 +02:00
Wim Taymans
cb344828a4 avidemux: don't push EOS in streaming mode 
In streaming mode, avidemux is not supposed to send an EOS event downstream but
it is supposed to return UNEXPECTED from the chain function instead so that
upstream can do the right EOS handling.
 2009-04-14 17:27:05 +02:00
Sebastian Dröge
108774781d Add initial support for muxing/demuxing Speex audio 
Note: This is not in the Matroska spec yet
Fixes bug #578310.
 2009-04-13 14:03:03 +02:00
Wim Taymans
776b0ae8cb pulsesink: handle NULL timing info 
Don't crash when the timing info is not yet available.
 2009-04-10 21:32:54 +02:00
Stefan Kost
b3d66d5e8d pulse: make it work on 0.9.12 
First we ignore request to fill the ringbuffer which are less then a segment.
The small request where causing stutter.
Then we disable flushing the stream when running against pa 0.9.12 as this
triggers an assertiong in the sound server and terminates it. It does not happen
with 0.9.10 and 0.9.14.
 2009-04-10 21:42:13 +03:00
Wim Taymans
963b343548 pulsesink: handle server disconnect in get_time 
When the server is disconnected or when we are shut down, make our clock return
an invalid time instead of erroring out.
 2009-04-10 14:18:48 +02:00
Wim Taymans
20a6908dfd pulsesink: bps is signed int to avoid overflow 
Keep bps as gint instead of guint because we will be doing signed math with it
later on and we don't want weird results.
 2009-04-10 12:01:27 +02:00
LRN
3e7aede3ea avidemux: add convert query, fix duration query 
Fix the duration query so that it also works with formats other than
TIME, such as DEFAULT to get the number of frames.

Add a convert function.

Fixes #578052.
 2009-04-10 00:26:44 +02:00
Wim Taymans
7d438518fb pulsesink: check for a stream 
Don't try to change the stream volume (and other things) when we don't have a
stream yet. Just store the values for later.
 2009-04-09 23:43:58 +02:00
Wim Taymans
ae83945349 pulsesink: fix compilation for newer pulseaudio 2009-04-09 18:07:38 +02:00
Wim Taymans
8d58de128d pulsesink: uncork fixes and use prebuf = 0 
We can use prebuf = 0 to instruct pulse to not pause the stream on underflows.
This way we can remove the underflow callback. We however have to manually
uncork the stream now when we have no available space in the buffer or when we
are writing too far away from the current read_index.
 2009-04-09 17:26:21 +02:00
Wim Taymans
d849340e64 pulsesink: handle write errors 2009-04-09 17:26:20 +02:00
Wim Taymans
81c5fb9e48 pulsesink: write silence on underflow 
Start filling up the buffer with empty samples when an underflow happens. We
need to do this to keep pulseaudio reporting the right time for us.
 2009-04-09 17:26:20 +02:00
Wim Taymans
2e2f1d73ca pulsesink: handle pull-based scheduling 
Use the default basesink methods for implementing pull based scheduling, it
works fine for us.
 2009-04-09 17:26:20 +02:00
Wim Taymans
8855ed90c0 pulsesink: add beginnings of pull-based scheduling 2009-04-09 17:26:20 +02:00
Wim Taymans
236baa5a13 pulsesink: keep track of clock reset 
when we switch streams, the clock will reset to 0. Make sure that the provided
clock doesn't get stuck when this happens by keeping an initial offset. We also
need to make sure that we subtract this offset in samples when writing to the
ringbuffer.
 2009-04-09 17:26:20 +02:00
Wim Taymans
6bc6cafcc6 pulsesink: rewrite pulsesink 
Derive from BaseAudioSink and implement our custom ringbuffer that maps to the
internal pulseaudio ringbuffer.
 2009-04-09 17:26:20 +02:00
Wim Taymans
28d733d53b pulse: remove some stray debug lines 2009-04-09 17:26:20 +02:00
Tim-Philipp Müller
e14bae6637 jpegdec: use slightly more adaptive formula for QoS 
Should work at least a tad better if the decoder can't keep up, and
should also spread dropped frames a bit more evenly over time.
 2009-04-09 11:34:19 +01:00
Stefan Kost
1095e624ec wavparse: don't leak pad-template 
gst_element_class_add_pad_template() does not take ownership.
 2009-04-07 22:36:39 +03:00
Felipe Contreras
c4c5de6044 Automatic update of common submodule 
From d0ea89e to b3941ea
 2009-04-04 21:18:55 +03:00
Thomas Vander Stichele
8009fcf547 add pending_samples so that we only update segment's last stop after really sending the samples 2009-04-04 15:14:32 +02:00
Thomas Vander Stichele
5f802dad4e add debug and an assert 2009-04-04 15:14:31 +02:00
Thomas Vander Stichele
fb4953a68d add debugging 2009-04-04 15:14:31 +02:00
Thomas Vander Stichele
be94a147ba add a test to check that we get all decoded bytes 
from a 10-buffer audiotestsrc flac, in the case of:
 - a full decode
 - a decode of a seek for the full file
 - a decode of a seek for a small part, smaller than the first buffer

The test fails because flacdec drops the first outgoing buffer on a seek
 2009-04-04 15:14:31 +02:00
Thomas Vander Stichele
5e19fc1058 clipping should also work if it's done on the first buffer starting at 0 2009-04-04 15:14:31 +02:00