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()
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.
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.
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.
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).
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.
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.
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.
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.
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
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()
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.
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.
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.
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.
Free the key value before we remove the header item from the array. The item we
retrieved from the array is only valid until we remove it from the array.
Save the tunnelid in the connection. Add a method to retrieve the tunnelid so
that a server can store and match the id against other tunnel requests.
Fix the URI in the tunnel requests so that they contain the absolute uri and the
query string if any instead of just the hostname.
Transparently base64 decode the input stream when tunneling.
Add method to set the connection ip address so that it can be included in the
tunnel response.
Add method to connect the two tunnel requests.
Add two callbacks for the async mode to notify a tunnel start and tunnel
complete event.
Add method to reset the watch after the connection has been tunneled.
Various little refactoring to make more stuff reusable.
API: RTSP::gst_rtsp_connection_set_ip()
API: RTSP::gst_rtsp_connection_get_tunnelid()
API: RTSP::gst_rtsp_connection_do_tunnel()
API: RTSP::gst_rtsp_watch_reset()
Add support for tunneling RTSP over HTTP.
Fix documentation some more.
See also #573173.
API: RTSP:gst_rtsp_connection_is_tunneled()
API: RTSP:gst_rtsp_connection_set_tunneled()
Add transport define for RTSP tunneled over HTTP.
Parse rtsph:// uris as tunneled HTTP over TCP.
API: GstRTSPLowerTrans::GST_RTSP_LOWER_TRANS_HTTP
See also #573173.
Add gst_rtsp_connection_get_url() method.
Reserve space for 2 sockets, one for reading and one for writing. Use socket
pointers to select the read and write sockets. This should allow us to implement
tunneling over HTTP soon.
API: RTSP::gst_rtsp_connection_get_url()
The previous change to appsrc/appsink requires people to 'make clean'
to get the marshallers rebuilt (causing a build failure otherwise).
Change some lines in the .list file around to force a rebuild of
these files automatically.
Add a uri handler to appsink.
don't emit signals when we have installed callbacks on appsink.
Add callbacks to appsrc to replace the signals.
Add property to disable callbacks in appsrc, default to TRUE for backwards
compatibility but disable when callbacks are installed.
API: GstAppSrc::emit-signals
API: GstAppSrc::gst_app_src_set_emit_signals()
API: GstAppSrc::gst_app_src_get_emit_signals()
API: GstAppSrc::gst_app_src_set_callbacks()
Add some padding to the callbacks structure just to be safe.
Remove the now invisible marshaller methods from the docs.
Fix a comment in the unit test.
Add a .def file for win32 builds (and make check-exports).
Fix LDFLAGS in Makefile.am, so the usual export regexps are used (fixes#573165).
Make sure private marshaller functions aren't exported by prefixing them with __gst;
also rename gst_app_marshal_OBJECT__VOID to _BUFFER__VOID, make it static and add
a comment why we're not using glib-genmarshal for this one.
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
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.
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).
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.
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()
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()
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.
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().
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.
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.
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()
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.
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>
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).
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.
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.
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.
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().
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.
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.
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.
