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
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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
...
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
...
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
...
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
Jonathan Matthew
9b7c9208c3
id3v2mux: write RVA2 frames containing peak/gain volume data
2009-04-02 00:20:02 +01:00
Tim-Philipp Müller
f1fb1f80fa
jpegdec: demote some log message from DEBUG to LOG
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And log decoder object.
2009-04-02 00:05:14 +01:00
Tim-Philipp Müller
00c4b0b17a
jpegdec: implement basic QoS
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Don't decode frames that are going to be too late anyway.
2009-04-01 21:15:02 +01:00
Tim-Philipp Müller
cb15d09c4a
rtspsrc: don't emit ugly warnings with older rtpjitterbuffer versions
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The on-npt-stop signals was added only recently to rtpjitterbuffer in
-bad, so check if the signal exists before g_signal_connect()ing to
it, to avoid warnings.
2009-04-01 12:29:33 +01:00
Wim Taymans
b037369d5b
rtspsrc: add proxy support
2009-03-31 19:08:37 +02:00
Stefan Kost
605ded5292
matroska: don't leak serialized values when writing tags
2009-03-31 17:16:04 +03:00
Stefan Kost
5ac6b84475
matroska: don't alter passed data and especialy don't leak.
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If we need different size, Make a copy, work with that and free it.
2009-03-31 17:06:50 +03:00
Stefan Kost
fa8e2d9bfe
goom: the structure is not fully initialized, but the copied.
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Set to fully to 0 to avoid creep of uninitialized values.
2009-03-31 16:42:15 +03:00
Stefan Kost
ef7bcf7bd1
matroska: init endianess as such and signedness as boolean.
2009-03-31 16:25:58 +03:00
Stefan Kost
0889ac1092
qtdemux: don't use ininitialized var in debug log statement
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Also make the log statement useful by printing the human readable format name.
2009-03-31 16:22:42 +03:00
Stefan Kost
f4f6d9799c
qtdemux: don't leak atom data in case of a wrong fourcc
2009-03-31 12:01:21 +03:00
Stefan Kost
9b8f1cbaa2
matroska: don't leak read data in demuxer
2009-03-31 11:57:36 +03:00
Stefan Kost
ba2c101963
udp: don't use protocol in debug message after freeing
2009-03-31 11:50:41 +03:00
Tim-Philipp Müller
6e5f789fa0
rtpmp4adepay: output should be framed already
2009-03-30 14:13:29 +01:00
Tim-Philipp Müller
11a8aa91b8
flac: require a 'newer' flac and remove support for the legacy flac API
2009-03-27 21:27:30 +00:00
Wim Taymans
fd18185d44
rtspsrc: link to the on_npt_stop signal to EOS
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Connect to the on_npt_stop signal of the session manager to schedule the EOS
actions.
2009-03-27 17:49:15 +01:00
Mark Nauwelaerts
3360f449c0
qtdemux: some stream synchronization to aid seeking in unbalanced clips
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Some clips (trailers) may have (length-wise) unbalanced streams,
which stalls the pipeline if seeking into that region.
Additional stream synchronization can handle this, as well as
sparse (subtitle) streams (at some later time ?)
2009-03-26 14:39:06 +01:00
Mark Nauwelaerts
a5502c9b37
qtdemux: additional safety and sanity checks (push based mode)
2009-03-26 14:38:02 +01:00
Wim Taymans
07329bc083
videomixer: some more indent fixes
2009-03-26 10:18:31 +01:00
Wim Taymans
62d118678a
videomixer: fix gst-indent screwup
2009-03-26 10:17:48 +01:00
Tim-Philipp Müller
37634c2afb
rtspsrc: better error message when the RTSP extension for Real streams is missing
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Try to post a decent error message when it looks like we're failing
because the Real RTSP extension plugin is missing. Also add i18n
bits for rtspsrc so our error messages get translated.
2009-03-25 17:54:35 +00:00
Tim-Philipp Müller
2199592039
i18n: make sure gettext gives us UTF-8 at all times
2009-03-25 15:42:15 +00:00
Tim-Philipp Müller
4bcd50ccbd
rtpmp4apay,rtpmp4depay: fix buffer leaks in AAC payloader and depayloader
2009-03-25 10:36:49 +00:00
