Wim Taymans
457d39075c
rtp: cleanups, add _list_get_seq() too
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Clean up the docs a little.
Add missing _list_get_seq method.
Add new symbols to the docs
2009-06-18 19:04:52 +02:00
Wim Taymans
e2ccc1ee39
rtp: cleanups
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Add Since tags to docs
Move some code around
Add win32 symbols
2009-06-18 18:51:04 +02:00
Wim Taymans
66c388a0e0
rtp: add bufferlist support
2009-06-18 18:51:04 +02:00
Wim Taymans
f385081c92
rtp: pass data to macros instead of GstBuffer
2009-06-18 18:50:35 +02:00
Peter Kjellerstedt
4fd61fbaa4
rtsp: Made the parsing of the RTSP URL scheme more generic.
2009-06-17 18:34:57 +02:00
Peter Kjellerstedt
726a47f777
rtsp: Added gst_rtsp_watch_queue_data().
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gst_rtsp_watch_queue_data() is similar to gst_rtsp_watch_queue_message()
but allows for queuing any data block for writing (much like
gst_rtsp_connection_write() vs. gst_rtsp_connection_send().)
API: gst_rtsp_watch_queue_data()
2009-06-17 18:34:33 +02:00
Peter Kjellerstedt
595f8b6d00
rtsp: Only extract the session ID from RTSP responses.
2009-06-17 18:02:18 +02:00
Peter Kjellerstedt
ddbeb44f14
rtsp: Added support for parsing IPv6 addresses in RTSP URLs.
2009-06-17 18:00:17 +02:00
Peter Kjellerstedt
95a606a0bb
rtsp: Use getaddrinfo() to support both IPv4 and IPv6.
2009-06-17 17:59:47 +02:00
Peter Kjellerstedt
e1a4c8871a
rtsp: Improved base64 decoding in fill_bytes().
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The base64 decoding in fill_bytes() expected the size of the read data to
be evenly divisible by four (which is true for the base64 encoded data
itself). This did not, however, take whitespace (especially line breaks)
into account and would fail the decoding if any whitespace was present.
2009-06-17 17:53:54 +02:00
Wim Taymans
ffd90dda89
audiosrc: fix get_offset
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When we need to jump to the most recently captured sample, jump to where the
next sample will be written instead of to some old data.
Fixes #581460
2009-06-17 14:00:23 +02:00
Wim Taymans
57a13f28de
audiosink: free the ringbuffer when going to NULL
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Unparent and free the ringbuffer when going to NULL, like we do with the
audiosrc element. We can do this now because we correctly manage the time
jumping back to 0.
2009-06-17 13:18:18 +02:00
Wim Taymans
e4492c24ea
audio: correctly handle short read/writes
2009-06-17 13:17:30 +02:00
René Stadler
2c5f455423
baseaudiosrc: add some extra logging for buffer timestamps
2009-06-17 12:36:50 +02:00
Sebastian Dröge
a64caea0bd
videofilter: Add a default get_unit_size function
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This returns the correct values for all formats that are handled by
GstVideoFormat and makes all the custom get_unit_size functions in
many elements unnecessary.
2009-06-16 19:38:17 +02:00
Wim Taymans
33837d420c
rtsp: add Timestamp header field
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fixes #585994
2009-06-16 18:57:20 +02:00
Tim-Philipp Müller
70089160f8
audiosink, audiosrc: do the class_ref()s in the right class_init functions
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Spotted by Philip Jägenstedt. Hopefully fixes #585970 for real.
2009-06-16 14:14:26 +01:00
Tim-Philipp Müller
3767cb6005
audiosink,audiosrc: ref the audio ring buffer class and type in class_init
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Hack around thread-safety issues in GObject and our racy _get_type()
functions (we could easily fix the _get_type() functions, but we still
need to hack around the GObject class races until we require a newer
GLib version, I think).
2009-06-15 15:39:09 +01:00
Wim Taymans
a5491ba218
audiosrc: return FALSE when receiving a SEEK event
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When receiving a seek event, return FALSE as we don't implement seeking.
2009-06-15 12:57:39 +02:00
Peter Kjellerstedt
73dd8236ce
rtsp: Use a more consistent naming of GstRTSPRec variables.
2009-06-15 09:28:34 +02:00
Peter Kjellerstedt
ff38999c8b
rtsp: Call message_sent() callback for all sent messages.
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Previously the messages_sent() callback was only called for messages
which had a CSeq, which excluded all data messages. Instead of using the
CSeq as ID, use a simple index counter.
2009-06-15 09:28:13 +02:00
Wim Taymans
a9c82f9472
ringbuffer: handle border cases in resampler
2009-06-11 19:13:28 +02:00
Wim Taymans
8bbf2e8a32
docs: fix typo
2009-06-11 12:39:19 +02:00
Wim Taymans
69b7fb3845
baseaudiosink: reset accum when dropping samples
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When we are resampling and we drop samples because we paused, reset the accum
counter because it's now invalid.
2009-06-11 12:38:35 +02:00
Jan Schmidt
c1bc55a4f5
docs: Fix a couple of warnings from the docs build.
2009-06-11 11:16:15 +01:00
Tim-Philipp Müller
249d9b4aa1
Don't include config.h multiple times when build audio testchannel app.
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Fixes build problem on win32 (#585075 ).
2009-06-10 21:37:29 +01:00
Wim Taymans
e01fab3ace
rtsp: add some more docs
2009-06-09 22:00:53 +02:00
Peter Kjellerstedt
263c5b227b
rtsp: Avoid a compiler warning.
2009-06-09 18:24:55 +02:00
Peter Kjellerstedt
dfc57e3f8a
rtsp: Updated documentation for GstRTSPResult.
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Moved GST_RTSP_ELAST to be last in the documentation to match the actual
enum values.
2009-06-09 18:23:28 +02:00
Peter Kjellerstedt
9c40eeeb4c
rtsp: Plug a memory leak.
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Free memory related to any partially read and/or written RTSP messages.
2009-06-09 16:28:20 +02:00
Wim Taymans
38e59ec75d
baseaudiosink: no need to cause discont when clipping
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Remove the discont-when-clipping hack now that basesink provides us with
correctly clipped samples when stepping.
2009-06-09 12:09:15 +02:00
Wim Taymans
cb4952fc2e
audiosink: don't align when we clip
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Don't align samples when they were clipped. Not entirely correct but better than
nothing for now.
2009-06-08 17:26:59 +02:00
Edward Hervey
ee3b251234
pbutils: Add description for hdv/aux-* formats.
2009-06-08 10:25:00 +02:00
Tim-Philipp Müller
5da78c8489
libgsttag: don't extract genres from empty ID3v1 tags
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If we don't have any other info, don't try to interpret the
genre field. In particular we don't want to interpret a genre
of 0 as 'Blues' if no other fields are set and the entire tag
is just empty.
2009-06-06 12:04:12 +01:00
Peter Kjellerstedt
2dbd8702dd
rtsp: Fixed a typo.
2009-06-05 14:06:17 +02:00
Peter Kjellerstedt
de18ad458f
rtsp: Remove an unused variable.
2009-06-05 14:05:54 +02:00
Peter Kjellerstedt
b0a9848524
rtsp: Removed duplicate initialization of conn->writefd.
2009-06-05 13:59:14 +02:00
Peter Kjellerstedt
0167e3589d
rtsp: Use #defined status codes.
2009-06-05 13:55:08 +02:00
Peter Kjellerstedt
c1a6644a18
rtsp: Correct gen_tunnel_reply().
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Prevent gen_tunnel_reply() from generating an incomplete response
in case an error response code is given.
2009-06-05 13:53:29 +02:00
Wim Taymans
59d9833924
rtsp: add G_LIKELY because we can
2009-06-02 12:10:39 +02:00
Peter Kjellerstedt
d8e0b5a4da
rtsp: Avoid compiler warnings with -Wextra.
2009-06-01 09:59:22 +02:00
Peter Kjellerstedt
848b834cb9
rtsp: Include gst/gstconfig.h to make sure GST_PADDING is defined.
2009-06-01 09:58:27 +02:00
Peter Kjellerstedt
e69c3a4f70
sdp: Remove an unused variable.
2009-06-01 09:43:04 +02:00
Wim Taymans
dcc42d5f92
netbuffer: also note the order of IP4 addresses
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IP4 addresses are also stored in network byte order. Make a note of this in the
docs.
2009-05-27 11:08:37 +02:00
Tim-Philipp Müller
6292ff4ae0
Revert "rtspconnection: don't use GLib-2.16 API, we require only 2.14"
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This reverts commit 418760cf74
2009-05-26 18:21:31 +01:00
Wim Taymans
796f8e2f76
netbuffer: document that the port is network order
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Document the fact that we store the port number in network order in
GstNetAddress and that the caller should byteswap appropriately.
2009-05-26 15:39:18 +02:00
Andy Wingo
c7ca6abe53
add can-activate-pull property to baseaudiosink
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* gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
to baseaudiosink.
2009-05-26 13:17:44 +02:00
Bastien Nocera
9c508ba458
cddabasesrc: Remove copy of sha1 digest
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Remove our copy of sha1 digest now that we depend on glib 2.16.
Fixes #536313
2009-05-26 11:11:03 +02:00
Tim-Philipp Müller
5fa9a8f4d0
video: don't expose internal gst_adapter_get_buffer() helper function
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If it's really needed it should go into GstAdapter in core.
2009-05-25 00:19:25 +01:00
David Schleef
538c1cde31
basevideo: Fix memleak
2009-05-22 21:29:51 -07:00
