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1461 commits

Author SHA1 Message Date
Stefan Kost
57a7d6f699 docs: add basic section docs for multichannel and relocate the ones for audio 
Add section docs for multichannel, so that it has a short desc in the toc too.
Move the section docs in adio up, so that the follow the copyright like
elsewhere.
 2009-06-27 23:25:09 +03:00
Руслан Ижбулатов
07c237ad19 Define WINVER before including any win headers 
Fixes bug #587080.
 2009-06-27 14:02:50 +02:00
René Stadler
41b7504e9c riff: prevent crash if rounded up tag size exceeds data size 
When rounding up `tsize' exceeds the remaining buffer size, `size' underflows
and an invalid read past the buffer data follows.
 2009-06-27 01:22:52 +03:00
Sebastian Dröge
939baee2bd basevideocodec: By default don't allow caps changes on the srcpad 
This fixed playback of Dirac files with schrodec when upstream wants
a different width/height, basevideocodec accepts this and then
pushes buffers with new caps but content of the old caps.
In the best case this will just result in wrong unit size and a
failure in basestransform elements.
 2009-06-26 15:20:09 +02:00
Tim-Philipp Müller
adff66fc83 pbutils: add description for multipart 
So we get slightly nicer error messages when multipartdemux is missing.
 2009-06-24 09:51:11 +01:00
Wim Taymans
85af9b82e8 basertppayload: add support for bufferlists 
Based on patch from Ognyan Tonchev.

See #585559
 2009-06-19 15:52:34 +02:00
Wim Taymans
f5c8055edf rtpbuffer: use new convenience functions 
New core convenience functions makes the list getters and setters trivial.
Maybe even too trivial...
 2009-06-19 15:33:04 +02:00
Wim Taymans
457d39075c rtp: cleanups, add _list_get_seq() too 
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 
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(). 
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(). 
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 
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 
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 
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 
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 
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 
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 
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. 
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 
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. 
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. 
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. 
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 
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 
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 
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(). 
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 
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" 
This reverts commit 418760cf74.

We now require GLib 2.16.
 2009-05-26 18:21:31 +01:00
Wim Taymans
796f8e2f76 netbuffer: document that the port is network order 
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 
* 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 
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 
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
David Schleef
35aae561e8 basevideo: Add preset interface to encoder 2009-05-22 17:34:56 -07:00
Wim Taymans
81170c4989 audiosink: improve debug message 2009-05-21 10:48:49 +02:00
Michael Smith
35a9de28f4 gstid3tag: Don't extract a track number unless present. 
In ID3v1, a track number is present only if byte 125 is null AND
byte 126 is non-null. If the track number is not present, don't add
a track number tag with value 0.
 2009-05-19 18:12:18 -07:00
Wim Taymans
243d366b34 videoutils: remove adapter methods 
Remove adapter methods now that they are in core.
 2009-05-20 00:48:40 +02:00
Wim Taymans
c68a361e31 audiosink: return the return value of wait_preroll 
Return the value that _wait_preroll() returned instead of always WRONG_STATE.
 2009-05-19 17:17:37 +02:00
David Schleef
17f3810f7b video: remove // comments 2009-05-15 16:21:15 -07:00
David Schleef
45cf881f39 video: Add Y444, v210, v216 formats 2009-05-15 16:18:59 -07:00
David Schleef
4ec34e83d5 video: Copy BaseVideo classes from Schroedinger 2009-05-15 16:18:58 -07:00
Tim-Philipp Müller
f2031e1313 pbutils: add descriptions for 3GP, JPEG 2000 and Motion JPEG 2000 2009-05-15 20:50:06 +01:00
Wim Taymans
b9723f6e1c audioclock: make our internal time monotonic 
Make the internal time increase monotonically.
 2009-05-13 21:38:56 +02:00
Sebastian Dröge
ab75db1653 propertyprobe: Fix typo in the docs 2009-05-12 15:53:07 +02:00
Wim Taymans
0a09632396 rtpdepay: add some more comments 2009-05-12 10:39:49 +02:00
Wim Taymans
d655120ee6 audioclock: make sure values are ever increasing 2009-05-12 10:39:41 +02:00
Sebastian Dröge
24dd91b1f0 interfaces: Seperate some more struct definitions from typedefs 2009-05-12 09:03:25 +02:00
Sebastian Dröge
e057414049 interfaces: Seperate some more struct definitions from typedefs 2009-05-12 09:03:25 +02:00
Sebastian Dröge
59aa1251d9 interfaces: API: Add gst_mixer_get_mixer_type() 
This is a convenience function that returns the mixer_type
of the interface struct.
 2009-05-12 09:03:24 +02:00
Sebastian Dröge
29b063b39b interfaces: Add docs for gst_color_balance_get_balance_type() 2009-05-12 09:03:24 +02:00
Sebastian Dröge
9fc4d195e1 vorbistag: Use text/uri-list as mimetype instead of ---> for URI lists 2009-05-12 09:03:22 +02:00
John Millikin
ef473dd0ae vorbistag: Store cover art in vorbiscomments 
Fixes bug #513373.
 2009-05-12 09:03:22 +02:00
Sebastian Dröge
e1875bf25f interfaces: API: Add gst_color_balance_get_balance_type() 
This is a convenience function that returns the balance_type
of the interface struct.
 2009-05-12 09:03:22 +02:00
Sebastian Dröge
b6c3567b41 interfaces: Separate struct definitions from typedefs 2009-05-12 09:03:22 +02:00
Tim-Philipp Müller
279b996d20 pbutils: add description for APE tag caps 2009-05-12 01:59:01 +01:00
Tim-Philipp Müller
3d33e2a873 tagdemux: cache events from upstream and re-send them once we have a source pad 
Makes sure tags don't get dropped when we have multiple tag demuxers in a row.
Fixes #580318.
 2009-05-12 01:15:21 +01:00
Michael Smith
8f6399f109 riff: support UYVY raw 4:2:2 in riff. 2009-05-11 14:04:16 -07:00
Andy Wingo
9f74ce745f Revert "add can-activate-pull property to baseaudiosink" 
This reverts commit c4074a2ee4.
 2009-04-29 11:18:42 +02:00
Andy Wingo
219a31fa3c Revert "[baseaudiosink] add docs for can-activate-pull" 
This reverts commit 416ce16f26.
 2009-04-29 11:18:33 +02:00
Andy Wingo
416ce16f26 [baseaudiosink] add docs for can-activate-pull 
* gst-libs/gst/audio/gstbaseaudiosink.c: Add documentation for
  can-activate-pull.
 2009-04-28 18:48:33 +02:00
Andy Wingo
c4074a2ee4 add can-activate-pull property to baseaudiosink 
* gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
  to baseaudiosink.
 2009-04-28 18:28:50 +02:00
Tim-Philipp Müller
8efe6108c4 cddabasesrc: fix posting of discid tags after MERGE_MODE_REPLACE_ALL changes in core 
Don't use REPLACE_ALL merge mode when that's not really what we want,
as now that REPLACE_ALL actually does what it's supposed to do in
core, we drop tags we wanted to keep, such as the various disc id
tags. Add unit test for this as well. Fixes #579463.
 2009-04-19 18:15:28 +01:00
Tim-Philipp Müller
418760cf74 rtspconnection: don't use GLib-2.16 API, we require only 2.14 
Fixes #579267.
 2009-04-17 10:35:34 +01:00
Wim Taymans
32904de58f baseaudiosink: don't unparent the ringbuffer 
when going to NULL, don't unparent the ringbuffer because we don't support going
back to 0 very well yet.
Fixes #579203
 2009-04-17 11:03:32 +02:00
Olivier Crete
d927114ef8 RTCP: don't fail when retrieving invalid PT 
We can't meaningfully assert on valid packet types so just return the type as it
is. Update the comments to reflect this.

Fixes #579192.
 2009-04-17 10:53:10 +02:00
Wim Taymans
f83f57b648 app: add trivial cast macros 
Add trivial cast macros for appsrc and appsink. Mark them as being since 0.10.23
and add the macros to the standard macros in the docs.

Fixes #579130
 2009-04-16 12:14:43 +02:00
Sebastian Dröge
a6cf0c8f06 video: Fix typo in the docs 2009-04-15 15:35:59 +02:00
Sebastian Dröge
a1d8cfde9d video: Add support for YVYU YUV colorspace 2009-04-15 14:53:47 +02:00
Tim-Philipp Müller
75acca2835 docs: fix hyperlink and move fft attribution to the right place 2009-04-15 00:19:19 +01:00
Stefan Kost
ab24d9d65c log: use G_GUINT64_FORMAT instead of llu 2009-04-15 00:02:39 +03:00
Josep Torra
71ab187355 RTSP: add missing headers for WMS RTSP 
Add missing headers related to Windows Media RTSP extension.
Fixes #578942
 2009-04-14 18:31:52 +02:00
Tim-Philipp Müller
9f23b82b2c Give credit to Mark Borgerding (kissfft author) 
and add myself to AUTHORS as well. Fixes #575638.
 2009-04-14 17:11:19 +01:00
Johann Prieur
86edcadc43 RTCP: add beginnings of Feedback messages 
Add the beginnings of parsing and constructing Feedback messages.
Fixes #577610.
 2009-04-14 16:45:20 +02:00
Wim Taymans
dffd1bcc97 baseaudiosrc: adjust the internal timestamp 
Adjust the internal timestamp before comparing it against the adjusted clock
time.
Fixes #578506
 2009-04-14 13:16:14 +02:00
Wim Taymans
0c4c1410f9 baseaudiosink: use new clock time methods 
Use the unadjusted internal clock times to calculate the internal/external
offset when calibrating the clock.

When going to NULL, unparent and free the ringbuffer, like we do in the source
element.
See #578506
 2009-04-14 13:12:59 +02:00
Wim Taymans
4231d54823 audioclock: add methods for the internal offset 
Add two methods for getting the unadjusted time of the clock and one for
adjusting an internal time. We will need these methods for correctly handling
the time after a gst_audio_clock_reset().

Add a debug category and some debug lines to the audio clock.

API: gst_audio_clock_get_time()
API: gst_audio_clock_adjust()
API: GST_AUDIO_CLOCK_CAST()
 2009-04-14 13:08:52 +02:00