mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-08 18:39:54 +00:00
Code Issues 1 Projects Releases Wiki Activity
6577 commits 44 branches 1509 tags 454 MiB
Commit graph

6577 commits

This branch
This branch
All branches
Author SHA1 Message Date
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
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 
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 
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. 
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. 
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 
And log decoder object.
 2009-04-02 00:05:14 +01:00
Tim-Philipp Müller
00c4b0b17a jpegdec: implement basic QoS 
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 
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