Commit graph

1252 commits

Author SHA1 Message Date
Edward Hervey
e57073b6f9 Riff: Add fourcc for mpeg1-in-avi (as produced by mencoder) 2009-02-25 08:05:58 +01:00
Garret D'Amore
b8af1223db mixer interface: Add flags to enhance mixer interfaces
This patch adds a few flags to the mixer and mixerctrl interface to
better support OSSv4 (and potentially other backends).

Patch By: Garret D'Amore <garrett.damore@sun.com>
Signed-Off-By: Jan Schmidt <jan.schmidt@sun.com>

API: GST_MIXER_FLAG_HAS_WHITELIST, GST_MIXER_FLAG_GROUPING,
API: GST_MIXER_TRACK_NO_RECORD, GST_MIXER_TRACK_NO_MUTE,
API: GST_MIXER_TRACK_WHITELIST
2009-02-24 17:23:58 +00:00
Jan Schmidt
94791df88d rtsp: Fix a strict aliasing warning
Fix strict aliasing warnings from casting a sockaddr_storage and
using it as a sockaddr_in6. Use a union instead.
2009-02-24 16:49:40 +00:00
Wim Taymans
bb5e2d3f56 Match WSAStartup and WSACleanup correctly
Don't randomly call WSAStartup and WSACleanup but instead call the startup when
we create a connection and cleanup when we free it again. Because the internal
datastructure is refcounted, this should not cause any refcounting leaks when
the connection is managed correctly.
Fixes #562794.
2009-02-24 12:11:00 +01:00
Wim Taymans
6e560ae5d8 Add method for handling server requests
Add a receive_request so that extensions can react to server requests.
2009-02-23 10:57:08 +01:00
Sebastian Dröge
d659e8353d tagdemux: Unref the actual buffer instead of the memory address of the buffer 2009-02-22 19:12:00 +01:00
Edward Hervey
5ce5433152 libs/video: Fix gst_video_format_new_caps* functions.
Only add a 'interlaced=True' property to caps *IF* it is interlaced, else
don't add anything.
2009-02-22 13:42:33 +01:00
Wim Taymans
15cd839f81 Improve key/value parsing
Improve header field parsing by keeping a ref to the key/value instead of
copying it into a local variable.
2009-02-20 17:26:40 +01:00
Wim Taymans
bb4310203a Add trailing \0 to message length
We always put a trailing 0 at the end of the message body. Reflect this fact in
the length of the message.
2009-02-20 12:35:53 +01:00
Wim Taymans
0ffd5e703a Don't parse headers for data messages
Don't try to parse the headers on a data message because they don't have
headers.
2009-02-20 09:52:16 +01:00
Edward Hervey
a490b3f2dd video: Fix 'Since' tags 2009-02-19 17:40:45 +01:00
Edward Hervey
c44b067817 video: Add flags for interlaced video along with convenience methods for interlaced caps.
These three flags allow all know combinations of interlaced formats. They should
only be used when the caps contain 'interlaced=True'.

Fixes #163577 (yes, it's a 4 year old bug).
2009-02-19 16:11:44 +01:00
Wim Taymans
f187ffddce Make RTSPConnection opaque and rename RTSPChannel
Make the RTSPConnection object opaque so that we can extend it in the future.

Rename GstRTSPChannel to GstRTSPWatch to avoid confusing with the RTSP channels.
2009-02-19 15:55:07 +01:00
Edward Hervey
02f9079d6b Add some more mappings for h264 in riff 2009-02-19 13:24:39 +01:00
Wim Taymans
e5d8551552 Add method to install callbacks on appsink
Based on pacth by Martin Samuelsson <martin dot samuelsson at axis dot com>
Fixes #571299.

Add gst_app_sink_set_callbacks() to install a set of callbacks. This is a more
performant alternative to connecting to the signals.

Add a unit test for appsink.

Clean up some of the appsink docs.

API: GstAppSink::gst_app_sink_set_callbacks()
2009-02-19 10:44:31 +01:00
Wim Taymans
a2f04c8f61 Add RTSP accept method
Add a method to accept a connection on a socket and create a GstRTSPConnection
for it.

API: gst_rtsp_connection_accept()
2009-02-18 18:46:35 +01:00
Wim Taymans
a6d75bd33c Add RTSP channel object for async io
Add a GstRTSPChannel object that wraps a GSource around the RTSP connection so
that the connection can be monitored from a maincontext. This allows us to
operate in ASYNC mode, which is handy when building a server.

Rework the old code to use the async code under the hood.

API: gst_rtsp_channel_new()
API: gst_rtsp_channel_unref()
API: gst_rtsp_channel_attach()
API: gst_rtsp_channel_queue_message()
2009-02-18 17:42:59 +01:00
Tim-Philipp Müller
a624df17c4 tagdemux: don't abort when downstream pulls a buffer of size 0
Pulling a 0-sized buffer is allowed, and we should handle this correctly instead of
aborting. Fixes #571009 (wma file with ID3v2 tag).
2009-02-12 09:18:20 +00:00
Tim-Philipp Müller
1fedfec220 riff: error out on nonsensical chunk sizes instead of aborting
When encountering a nonsensical chunk size such as (guint)-1, error out cleanly instead of
continuing and trying to g_memdup() 4GB of data that doesn't exist, which will either abort
in g_malloc() or crash.

Fixes #553295, crash with fuzzed AVI file.
2009-02-11 16:58:18 +00:00
Peter Kjellerstedt
430eea3016 gstrtspmessage: Minor documentation correction.
Corrected documentation about what needs to be freed after calling
gst_rtsp_message_new(), gst_rtsp_message_new_request(),
gst_rtsp_message_new_response() and gst_rtsp_message_new_data().
2009-02-10 17:37:06 +01:00
Wim Taymans
76112f9f04 RTSPRange: Add method to serialize ranges
Add gst_rtsp_range_to_string() to serialize a GstRTSPRange to a string that can
be used by a server.
API: GstRTSPRange::gst_rtsp_range_to_string()
2009-02-04 17:03:52 +01:00
Wim Taymans
4bb5722f1a GstRTSPUrl: Add some const to methods
Add const to the methods that do not modify the object.
2009-02-04 13:16:48 +01:00
Wim Taymans
ad1dea3122 Add more g_return_if_fail() calls
Check that we have a valid file descriptor before entering certain functions in
order to avoid undesirable situations.
Add some more debugging in the connect method.
2009-02-04 11:18:31 +01:00
Tim-Philipp Müller
95d6fb0501 pbutils: remove duplicate detail strings when calling the external codec installer
It doesn't make sense to ask installers for the same codec or element twice, so filter out duplicate requests before calling the external helper script and make the unit test check this works right. Fixes #567636.
2009-02-02 17:34:23 +00:00
Stefan Kost
486fe43cb9 Add a FIXME 0.11. Make the log message a bit more detailed and add comments. 2009-02-02 18:05:42 +02:00
Wim Taymans
35cec4c006 Fix string leak in rtspmessage
when we remove a header field from a message we must free the value associated
with the key to avoid a memory leak.
2009-02-02 10:09:07 +01:00
Stefan Kost
950d0c0a7d Link to the class, as we can't link to the members yet. 2009-01-31 18:44:32 +02:00
Wim Taymans
6f3511bfb6 fix some typos
Fix some typos in the doc string of the new
gst_rtsp_options_as_string() method.
2009-01-29 14:00:30 +01:00
Wim Taymans
484a025f6d Add new RTSP message method to set header
Add gst_rtsp_message_take_header() that takes ownership of the passed header
value. This allows us to avoid an allocations and memory copy in some
situations.
API: GstRTSPMessage::gst_rtsp_message_take_header()
2009-01-29 11:55:10 +01:00
Wim Taymans
e8bd8cab41 Add method to serialize RTSP options
Add gst_rtsp_options_as_text() method to serialize a set of RTSP options to a
string.
API: GstRTSP::gst_rtsp_options_as_text()
2009-01-28 11:48:01 +01:00
Jan Schmidt
63c9ede3d0 Extend and clean up git ignores 2009-01-23 23:16:11 +00:00
Wim Taymans
a7f2540f77 Add more codec ids for RIFF formats
Handle codec ID for various other AAC formats.
Sync the list of possible codec ids with that of ffmpeg.
Fixes #567255
2009-01-23 11:33:29 +01:00
Wim Taymans
26256b95c8 Reset queued_bytes counter when flushing
Set the amount of queued bytes in the internal queue back to 0 when we clear the
queue.
Fixes #567982
2009-01-23 11:11:31 +01:00
Wim Taymans
509f561ef3 Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base 2009-01-22 13:12:02 +01:00
Sebastian Dröge
2e8f9921c9 Reduce the number of allocations for creating FFT contexts
Reduce the number of allocations from 2 to 1 for every FFT
context by allocating enough memory for the FFT context
and passing parts of it to the kissfft allocation functions.
2009-01-22 12:54:35 +01:00
Wim Taymans
9ce042e2a7 Avoid overflows in the padding checks by doing the check slightly
differently.
Add a unit test to check for correct behaviour.
2009-01-21 13:09:29 +01:00
Sebastian Dröge
4d3ff205be gst-libs/gst/fft/: Use correct struct alignment everywhere to prevent unaligned memory accesses, resulting in SIGBUS ...
Original commit message from CVS:
* gst-libs/gst/fft/_kiss_fft_guts_f32.h:
* gst-libs/gst/fft/_kiss_fft_guts_f64.h:
* gst-libs/gst/fft/_kiss_fft_guts_s16.h:
* gst-libs/gst/fft/_kiss_fft_guts_s32.h:
* gst-libs/gst/fft/kiss_fftr_f32.c: (kiss_fftr_f32_alloc):
* gst-libs/gst/fft/kiss_fftr_f64.c: (kiss_fftr_f64_alloc):
* gst-libs/gst/fft/kiss_fftr_s16.c: (kiss_fftr_s16_alloc):
* gst-libs/gst/fft/kiss_fftr_s32.c: (kiss_fftr_s32_alloc):
Use correct struct alignment everywhere to prevent unaligned
memory accesses, resulting in SIGBUS on sparc and probably others.
Fixes bug #500833.
2009-01-16 11:44:04 +00:00
Sebastian Dröge
98ea758763 gst-libs/gst/tag/gsttagdemux.c: Forward unknown events upstream to allow latency configuration.
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_srcpad_event):
Forward unknown events upstream to allow latency configuration.
Fixes bug #567960.
2009-01-16 11:40:02 +00:00
Jan Schmidt
80ac3b565e gst-libs/gst/app/gstappsink.c: Store the returned signal id in the right slot when registering the pull-buffer signal.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c:
Store the returned signal id in the right slot when
registering the pull-buffer signal.
Fixes #567168
Spotted by: Thomas Vander Stichele  <thomas at apestaart dot org>
2009-01-09 23:13:17 +00:00
Tim-Philipp Müller
d629c9fc17 gst-libs/gst/interfaces/mixer.c: Small docs addition to clarify that one really mustn't free the constant GList retur...
Original commit message from CVS:
* gst-libs/gst/interfaces/mixer.c:
Small docs addition to clarify that one really mustn't free
the constant GList returned (#566812).
2009-01-09 17:17:50 +00:00
Wim Taymans
1f6297f051 Add GType for GstRTSPUrl and expose a copy function because we can.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtsp/gstrtspurl.c: (register_rtsp_url_type),
(gst_rtsp_url_get_type), (gst_rtsp_url_copy):
* gst-libs/gst/rtsp/gstrtspurl.h:
* win32/common/libgstrtsp.def:
Add GType for GstRTSPUrl and expose a copy function because we can.
API: gst_rtsp_url_copy()
Fixes #567027.
2009-01-08 17:18:24 +00:00
Sebastian Dröge
ba03cb6080 gst-libs/gst/cdda/gstcddabasesrc.*: Make the GType of GstCDDABaseSrcMode public for bindings.
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
* gst-libs/gst/cdda/gstcddabasesrc.h:
Make the GType of GstCDDABaseSrcMode public for bindings.
Fixes bug #566837.
2009-01-07 10:50:15 +00:00
José Alburquerque
7431789249 gst-libs/gst/audio/gstaudioclock.*: Make gst_audio_clock_new use const gchar* to ease the wrapping of
Original commit message from CVS:
Patch by: José Alburquerque <jaalburqu svn gnome org>
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_new):
* gst-libs/gst/audio/gstaudioclock.h:
Make gst_audio_clock_new use const gchar* to ease the wrapping of
C++ bindings. Fixes #566723.
2009-01-06 17:30:31 +00:00
Tim-Philipp Müller
ada70bb159 gst-libs/gst/app/: Make debug categories static. Use _element_class_set_details_simple().
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsink.h:
* gst-libs/gst/app/gstappsrc.c:
* gst-libs/gst/app/gstappsrc.h:
Make debug categories static. Use _element_class_set_details_simple().
2009-01-06 11:10:29 +00:00
Tim-Philipp Müller
d2b82026c8 gst-libs/gst/app/: Move private data into a private instance struct. Add padding to instance and class structures exp...
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (_GstAppSinkPrivate),
(gst_app_sink_class_init), (gst_app_sink_init),
(gst_app_sink_dispose), (gst_app_sink_finalize),
(gst_app_sink_unlock_start), (gst_app_sink_unlock_stop),
(gst_app_sink_flush_unlocked), (gst_app_sink_start),
(gst_app_sink_stop), (gst_app_sink_event), (gst_app_sink_preroll),
(gst_app_sink_render), (gst_app_sink_getcaps),
(gst_app_sink_set_caps), (gst_app_sink_get_caps),
(gst_app_sink_is_eos), (gst_app_sink_set_emit_signals),
(gst_app_sink_get_emit_signals), (gst_app_sink_set_max_buffers),
(gst_app_sink_get_max_buffers), (gst_app_sink_set_drop),
(gst_app_sink_get_drop), (gst_app_sink_pull_preroll),
(gst_app_sink_pull_buffer)::
* gst-libs/gst/app/gstappsink.h: (GstAppSinkPrivate), (_GstAppSink)::
* gst-libs/gst/app/gstappsrc.c: (_GstAppSrcPrivate),
(gst_app_src_class_init), (gst_app_src_init),
(gst_app_src_flush_queued), (gst_app_src_dispose),
(gst_app_src_finalize), (gst_app_src_set_property),
(gst_app_src_get_property), (gst_app_src_unlock),
(gst_app_src_unlock_stop), (gst_app_src_start), (gst_app_src_stop),
(gst_app_src_is_seekable), (gst_app_src_check_get_range),
(gst_app_src_query), (gst_app_src_do_seek), (gst_app_src_create),
(gst_app_src_set_caps), (gst_app_src_get_caps),
(gst_app_src_set_size), (gst_app_src_get_size),
(gst_app_src_set_stream_type), (gst_app_src_get_stream_type),
(gst_app_src_set_max_bytes), (gst_app_src_get_max_bytes),
(gst_app_src_set_latencies), (gst_app_src_set_latency),
(gst_app_src_get_latency), (gst_app_src_push_buffer_full),
(gst_app_src_push_buffer_action), (gst_app_src_end_of_stream)::
* gst-libs/gst/app/gstappsrc.h: (GstAppSrcPrivate)::
Move private data into a private instance struct. Add padding to
instance and class structures exposed in public headers. Add
Since markers to the gtk-doc blurbs (#566750).
2009-01-06 10:56:45 +00:00
Jan Schmidt
1b2dc5f3a8 gst-libs/gst/video/: Fix up build flags and include statement for the new generated enumtypes files, to fix dist.
Original commit message from CVS:
* gst-libs/gst/video/Makefile.am:
* gst-libs/gst/video/video.h:
Fix up build flags and include statement for the new generated
enumtypes files, to fix dist.
2009-01-06 10:16:16 +00:00
Jan Schmidt
08393941a8 Move AppSrc/AppSink from gst-plugins-bad. Fixes #564421
Original commit message from CVS:
* configure.ac:
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
* docs/plugins/gst-plugins-base-plugins.interfaces:
* docs/plugins/gst-plugins-base-plugins.prerequisites:
* docs/plugins/gst-plugins-base-plugins.signals:
* docs/plugins/inspect/plugin-app.xml:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsrc.c:
* tests/examples/Makefile.am:
* tests/examples/app/Makefile.am:
Move AppSrc/AppSink from gst-plugins-bad. Fixes #564421
2009-01-05 23:04:57 +00:00
Wim Taymans
0a4c1bc64c gst-libs/gst/audio/gstbaseaudiosink.c: Avoid holding the OBJECT_LOCK when calling ringbuffer functions that take the ...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_change_state):
Avoid holding the OBJECT_LOCK when calling ringbuffer functions that
take the ringbuffer lock because rinbuffer lock > OBJECT_LOCK. We can do
this because the async_play method is deprecated and usually not called
anymore.
2009-01-05 17:13:13 +00:00
Edward Hervey
70a35897fb gst-libs/gst/tag/gsttagdemux.*: Add GType for GstTagDemuxResult enum.
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_result_get_type):
* gst-libs/gst/tag/gsttagdemux.h:
Add GType for GstTagDemuxResult enum.
2008-12-31 13:31:55 +00:00
Edward Hervey
98ad43fcdd gst-libs/gst/video/: Add glib-mkenum for GstVideoFormat enum GTYPE auto-generation.
Original commit message from CVS:
* gst-libs/gst/video/Makefile.am:
* gst-libs/gst/video/video.h:
Add glib-mkenum for GstVideoFormat enum GTYPE auto-generation.
This will help bindings to use it.
2008-12-31 13:01:30 +00:00