Commit graph

1422 commits

Author SHA1 Message Date
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
Wim Taymans
b674584e97 rtsp: fix g_return condition
when parsing a data message, we require a data message.
2009-03-11 17:29:41 +01:00
Wim Taymans
18f612ffa9 rtsp: free the right string.
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.
2009-03-11 14:09:54 +01:00
Wim Taymans
16225d45be rtsp: keep track of amount of decoded bytes
Keep track of the actual amount of decoded bytes, which can be less than 3 when
we decode the last bits of a base64 message.
2009-03-11 14:09:54 +01:00
Wim Taymans
f964c0fc38 rtsp: only add ports when not using TCP
Only add the port numbers in the transport string when we are using udp or
multicast.
2009-03-09 13:53:41 +01:00
Wim Taymans
bc54a5f9a0 rtsp: use gstreamer dump mem
--
2009-03-09 13:53:15 +01:00
Wim Taymans
3a72044a22 rtsp: use glib base64 encoder
--
2009-03-09 13:51:48 +01:00
Edward Hervey
a3c88fb32b Riff: Add mapping for Fraps video codec.
Found through insanity testrun. Confirmed mapping in libavformat.
2009-03-09 10:03:13 +01:00
Edward Hervey
b870b61c00 riff: Add the 'DVR ' mapping for mpeg2video.
Found this in 3 files from the insanity suite and mapping is also present
in libavformat.
2009-03-09 09:08:00 +01:00
LRN
eb3ff95a3a rtsp: fix compilation on windows.
Remove unused variable when building for windows.
Fixes #574443.
2009-03-08 18:17:48 +01:00
Wim Taymans
d998f6097b riff: add theora mapping
Add theora mappings. See #574169.
2009-03-06 18:54:57 +01:00
Wim Taymans
2cc1a6808d rtsp: Add methods for getting the read/write fds
API:gst_rtsp_connection_get_readfd()
API:gst_rtsp_connection_get_writefd()
2009-03-06 18:54:57 +01:00
Julien Moutte
d45b27d92d Fix build on Mac OS X 2009-03-06 10:37:38 +01:00
Wim Taymans
f69a3d953a rtsp: fix parsing of 'now-' ranges.
--
2009-03-05 13:48:37 +01:00
Wim Taymans
bcaec3d907 rtsp: do some more cleanup in _close
Do som more cleanup in gst_rtsp_connection_close() so that it's back into the
unconnected state as it was allocated.
2009-03-04 16:24:01 +01:00
Wim Taymans
629f2dcee4 rtsp: fix the memory management of the url
Constify the url parameter in _create.
Make a copy of the url stored in the connection.
Free the url when the connection is freed.
2009-03-04 16:11:20 +01:00
Wim Taymans
b6d7a1dc03 RTSP: Add support for server tunneling
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()
2009-03-04 12:21:29 +01:00
Wim Taymans
3b6e9fc870 rtsp: add new defines for tunneling
Add two more result codes for tunneling support.
2009-03-04 12:18:00 +01:00
Wim Taymans
9ea1240910 rtsp: remove , from last enum member
Remove , from last enum member to improve compatibility with other compilers.
2009-03-04 12:12:06 +01:00
Wim Taymans
9045d210b2 rtsp: remove debugging g_message
--
2009-03-02 16:13:33 +01:00
Wim Taymans
fbc4f2d4fe RTSP: add support for Quicktime tunneled RTSP
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()
2009-03-02 16:03:49 +01:00
Wim Taymans
40db590e71 RTSP: parse rtsph uris as RTSP tunneled over HTTP
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.
2009-03-02 15:48:56 +01:00
Wim Taymans
4664fe40bc rtsp: add _get_url method and separate sockets
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()
2009-03-02 10:58:49 +01:00
Tim-Philipp Müller
0a835bc9a3 app: force automatic rebuild of gstapp-marshal.[ch] after previous change
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.
2009-03-01 18:31:17 +00:00
LRN
e5d2d32bba rtspconnection: Use correct types for some functions on Win32
Fixes bug #573529.
2009-02-28 19:35:33 +01:00
Edward Hervey
ed013753c0 rtspconnection: Fix warning about using unitialized value. 2009-02-28 13:11:59 +01:00
Edward Hervey
6f73427aa6 riff: Add more codec mappings.
This comes mostly from a review of ffmpeg/libavformat/riff.c
2009-02-28 12:41:28 +01:00
Stefan Kost
4e4f922d7a rtsprange: don't leak the range in case of parsing error.
Free the gstRTSPTimeRange if we don't return it. Also simplify
gst_rtsp_range_free() as it is valid to pass NULL to g_free().
2009-02-26 18:01:05 +02:00
Wim Taymans
c4036dd701 app: add callbacks to appsrc, cleanups
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()
2009-02-26 16:44:53 +01:00
Wim Taymans
661f2da6e0 Appsink: add padding for callbacks + docs
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.
2009-02-26 11:42:44 +01:00
Stefan Kost
58695d78f9 docs: fix newly added interlace constants and plug holes in video format docs 2009-02-26 10:09:59 +02:00
Stefan Kost
251e4d160a docs: don't put random stuff in tags.
Tags like Since: or Returns: can only be followed by more tags. gtk-doc has no
tag to append text again to the documentation body.
2009-02-26 10:09:59 +02:00
Tim-Philipp Müller
07d2dbfdfe app: add win32 .def file and only export functions we want exported
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.
2009-02-25 19:50:00 +00:00
Peter Kjellerstedt
2fe8e4c1de Fixed a typo. 2009-02-25 16:25:33 +01:00
Peter Kjellerstedt
a038a8d46d rtsp, multifdsink: Unify the use of union gst_sockaddr. 2009-02-25 15:45:50 +01:00
Tim-Philipp Müller
3d88a5b985 riff: add fourcc for mpeg2-in-avi (as produced by mencoder)
Fixes: #565777
2009-02-25 11:13:01 +00:00
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