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
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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
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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
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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.
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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
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Fixes #579070 .
2009-04-15 19:30:11 +01:00
James Andrewartha
ac48c2d211
rtph263pay: fix compilation on big-endian
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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
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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.
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
Edward Hervey
5439fb89d1
Automatic update of common submodule
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From f8b3d91 to d0ea89e
2009-04-04 14:54:01 +02:00
Zaheer Merali
988a4c6532
Fix grammar.
2009-04-03 09:57:15 +01:00
Wim Taymans
b6bf3ba7d3
rtspsrc: allow http:// on the proxy setting
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Allow and ignore http:// at the start of the proxy setting, like
souphttpsrc.
Fixes #573173
2009-04-02 22:41:01 +02:00
Wim Taymans
40f6ed8875
rtspsrc: don't leak the udpsrc pad
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Fix memory leak in rtspsrc because we didn't unref the udpsrc pad.
See #577318
2009-04-02 21:08:48 +02:00
Michael Smith
85d7fb0599
rtptheorapay: fix length encoding in packed headers.
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As for vorbis payloader; this by inspection had the same bug.
2009-04-01 17:31:18 -07:00
Michael Smith
5f9d9e2243
rtpvorbispay: in packed headers, properly flag multibyte lengths.
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In the sequence of header lengths, for headers >127 bytes, we use
multiple bytes to encode the length. Bytes other than the last must have
the top (flag) bit set.
2009-04-01 17:23:33 -07:00