mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-07-27 00:38:21 +00:00
Code Issues 1 Projects Releases Wiki Activity
7077 commits 44 branches 1505 tags 340 MiB
Commit graph

1494 commits

Author SHA1 Message Date
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
Wim Taymans 251f152c20 baseaudiosink: use the internal clock time 
We can't assume that the internal clock time is the same as the function we
installed on our provided clock because somebody might have changed it.
 2009-04-10 21:50:55 +02:00
Martin Samuelsson ee03bf5379 appsink: make callbacks return GstFlowReturn 
Make the new_buffer and new_preroll callbacks return a GstFlowReturn so that
errors can be reported properly.
Fixes #577827.
 2009-04-09 23:46:17 +02:00
Wim Taymans e6798c5cce ringbuffer: allow for custom commit functions 
Allow subclasses to override the commit method.
 2009-04-09 18:04:44 +02:00
Wim Taymans cae2981f83 baseaudiosink: fix a small glitch after pause 
After we pause the stream and interrupt the writeout to the ringbuffer, also adjust
the amount of output samples we consumed. We can't do this reliably with the
current API when we are doing trick modes but we can do the right thing for
normal playback.
 2009-04-08 18:06:54 +02:00
Stefan Kost ff9ee1dc5a audiofilter: don't leak pad-template 
gst_element_class_add_pad_template() does not take ownership.
 2009-04-07 22:39:07 +03:00
Edward Hervey 2555eeb737 navigation/v4l: Don't use g_return_val_if_fail for computed/used values. 2009-04-04 16:28:14 +02:00
Wim Taymans 88110ea67e rtsp: use fully qualified urls when using a proxy 
Use a fully qualified url when specifying the url for tunneled requests through
a proxy.
See #573173
 2009-04-02 22:28:55 +02:00
Jan Schmidt 033e654172 navigation: Extend the navigation interface 
Add support for a set of standard commands that can be queried and executed to
support applications like DVD. Add query construction and parsing functions.
Add new messages that can be sent on the bus to provide notifications related
to commands, multiangle changes, and button highlight activity.
Add some helper functions to parse the existing GstNavigation events that
elements might receive.
Document it all and add unit tests.
 2009-04-02 12:21:18 +01:00
Wim Taymans eed784b372 rtsp: fix little typo in the comments 2009-04-01 09:03:35 +02:00
Tim-Philipp Müller fc8c5cba15 rtspconnection: make gst_rtsp_watch_queue_message() thread-safe 
People might queue messages from a thread other than the thread in which
the main context which this watch is attached is iterated from, so use
a GAsyncQueue instead of a GList, so g_list_append() doesn't trample
over list nodes just freed in the other thread. This just fixes issues
I've had with gst-rtsp-server. We might need more locking in various
places here.
 2009-03-31 18:30:57 +01:00
Tim-Philipp Müller dfe96ce618 rtsp: clear the entire builder structure 
And use structure instead of variable with sizeof when
clearing the rtsp message structure, for clarity.
 2009-03-31 18:30:48 +01:00
Tim-Philipp Müller dd9f077177 docs: fix typo in gst_rtsp_message_unset() API docs 2009-03-31 18:30:48 +01:00
Wim Taymans 8b37dc3eb8 rtsp: add support for proxies 
Add suport for proxy servers. Currently only used for tunneled HTTP
connections without authentication.
 2009-03-31 19:00:00 +02:00
Wim Taymans 8be68f983c Revert "rtsp: reset whole message (was sizeof pointer instead of sizeof type)" 
This reverts commit 79de0b8d67.
 2009-03-31 18:57:08 +02:00
Stefan Kost 79de0b8d67 rtsp: reset whole message (was sizeof pointer instead of sizeof type) 2009-03-31 12:27:09 +03:00
Jan Schmidt 43788e4796 doc: Fix a typo in the GstMixer docs 2009-03-31 00:58:24 +01:00
Wim Taymans 0d3d3026d2 rtsp: start CSeq counting from 1 instead of 0 
Start counting from 1 instead of 0 as this is what most other clients
seem to do.
 2009-03-25 16:37:28 +01:00
Wim Taymans 1081ae59eb rtsp: add ETag and If-Match headers 
Add new headers, we need them for RealMedia support.
 2009-03-25 16:36:14 +01:00
Tim-Philipp Müller 0267e79778 audiosrc: improve 'Dropped n samples' warning message 2009-03-25 11:27:44 +00:00
Sebastian Dröge 108ead73c8 rtsp: Use GLib base64 functions and deprecate gst_rtsp_base64_encode 
This also fixes another instance of CVE-2008-4316.
 2009-03-17 22:53:44 +01:00
Wim Taymans f4b7cbbf16 rtsp: fix resolving of hostnames 
We were returning a pointer to a stack variable with the resolved hostname,
which doesn't work.
return a copy of the resolved ip address instead.
Fixes #575256.
 2009-03-13 16:19:41 +01:00
Wim Taymans 91b2d71da0 appsrc: release lock in _eos flushing case 
Release the mutex when we are flushing in gst_app_src_end_of_stream()
Fixes #574964.
 2009-03-13 15:16:44 +01:00
Jan Schmidt 566583e871 vorbistag: Protect memory allocation calculation from overflow. 
Patch by: Tomas Hoger <thoger@redhat.com> Fixes CVE-2009-0586
 2009-03-12 15:02:07 +00:00
Wim Taymans 0e2157029e rtsp: fix parsing of the timeout parameter 
--
 2009-03-11 18:45:59 +01:00