Commit graph

14834 commits

Author SHA1 Message Date
Sebastian Dröge
4d1726fddd riff: Add missing closing parenthesis to GST_RIFF_WAVE_FORMAT_ANTEX_ADPCME
Apparently this #define is unused.
2016-01-03 10:33:53 +02:00
Stefan Sauer
f4ae53144e riff-ids: remove trailing whitespace 2016-01-02 23:29:43 +01:00
Stefan Sauer
adb24a54ca riff-ids: fix two swapped ids
For these fourcc ids the name and value is swapped. This was causing a warning
when registering the avi ids.
2016-01-02 23:29:43 +01:00
Sebastian Dröge
81cfb23945 sdp: Also reorder SUBDIRS to try even harder to build the RTP library first 2015-12-31 20:43:28 +02:00
Sebastian Dröge
bbd82057ab sdp: The SDP library depends on the RTP library now and is not independent anymore
Fix up the build dependencies.
2015-12-31 20:41:38 +02:00
Hyunjun Ko
682b523652 sdp: add helper fuctions from/to sdp from/to caps
<gstsdpmessage.h>
GstCaps*       gst_sdp_media_get_caps_from_media   (const GstSDPMedia *media, gint pt);
GstSDPResult   gst_sdp_media_set_media_from_caps   (const GstCaps* caps, GstSDPMedia *media);
gchar *        gst_sdp_make_keymgmt                (const gchar *uri, const gchar *base64);
GstSDPResult   gst_sdp_message_attributes_to_caps  (GstSDPMessage *msg, GstCaps *caps);
GstSDPResult   gst_sdp_media_attributes_to_caps    (GstSDPMedia *media, GstCaps *caps);

<gstmikey.h>
GstMIKEYMessage * gst_mikey_message_new_from_caps  (GstCaps *caps);
gchar *           gst_mikey_message_base64_encode  (GstMIKEYMessage* msg);

https://bugzilla.gnome.org/show_bug.cgi?id=745880
2015-12-31 17:11:57 +02:00
Sebastian Dröge
eb09889176 audioconvert: Pass pointer arrays instead of singleton pointers to gst_audio_converter_samples()
In this specific case it wouldn't cause problems as we only ever access the
first array element, but let's make explicit what is happening here.

CID 1346530 and 1346529
2015-12-29 18:14:54 +02:00
Sebastian Dröge
43655580e7 encoding-profile: Check for FALSE'ness directly, not by comparing with FALSE 2015-12-29 17:56:21 +02:00
Sebastian Dröge
f31240a765 encoding-profile: Don't use preset_name string after free
When we run the loop for another time and do not have a preset name, we would
try to print the preset name of a previous iteration that is already freed.

Also move some other variables into the block where they are actually used
to prevent similar mistakes in the future.

CID 1346536
2015-12-29 17:55:23 +02:00
Stefan Sauer
898940a37f audioconvert: add a test for gap handling 2015-12-29 14:40:32 +01:00
Stefan Sauer
7bbfa39ada audioconvert: fix passthrough operation
We did not take the sample size into account. Rearrange the tests to have more
conversion test and an extra test case for passthrough operations.

Fixes #759890
2015-12-29 14:40:32 +01:00
Tim-Philipp Müller
f2ecf85103 tools: gst-device-monitor: print uint properties in both decimal and hex
Some values are easier to read and make sense of in hex.

https://bugzilla.gnome.org//show_bug.cgi?id=759780
2015-12-29 11:29:31 +00:00
Reynaldo H. Verdejo Pinochet
e61f5b2138 videoblend: special case 1x1 src dims on increment computation
Fix crash with 1x1 overlay pixmap

https://bugzilla.gnome.org/show_bug.cgi?id=757290
2015-12-28 14:16:41 -08:00
Sebastian Dröge
0416f121f2 typefindfunctions: Make sure that enough data is available in AAC/ADTS typefinder
We would otherwise read beyond the array bounds and crash every now and then.
This was introduced with 5640ba17c8.

https://bugzilla.gnome.org/show_bug.cgi?id=759910
2015-12-28 13:51:02 +02:00
Stefan Sauer
267e7ba1d9 tests: remove commented code from audioconvert test
This is just what we have in gst_check_buffer_data().
2015-12-27 19:41:43 +01:00
Stefan Sauer
0bd3f818bb audio-converter: code cleanup
Rename samples to num_samples, since we also have samples in chain, but that is
the data pointer. Always use gzize for num_samples. Make the log output a bit
more homogenous.
2015-12-27 19:25:20 +01:00
Tim-Philipp Müller
69d3b098a2 tools: gst-device-monitor: print non-string device properties too 2015-12-26 11:43:22 +00:00
Sebastian Dröge
3459bd6854 audio: Fix some documentation warnings
Remove/rename function parameters and skip some functions that can't
be used by bindings as they are now.
2015-12-26 09:43:56 +01:00
Sebastian Dröge
3ba59f0b62 videoaffinetransformmeta: Add (transfer none) annotation for return value 2015-12-26 09:43:51 +01:00
Sebastian Dröge
6a57399270 playsink: Don't leak audio/video filters due to floating references weirdness
The filters' floating references are sinked during set_property() already,
which means that GstBin takes a new reference when adding the filter to it.
Get rid of the additional reference after adding the filter to the bin.
2015-12-25 11:34:10 +01:00
Sebastian Dröge
a136ac0e2f playsink: Allow reuse of audio/video filters by unparenting them from their bins
And also recreate the chains if the filter is changing.
2015-12-25 10:36:44 +01:00
Sebastian Dröge
24181db083 playsink: Don't leak audio/video filters when using non-raw media 2015-12-25 10:28:02 +01:00
Sebastian Dröge
7de6b06ade Back to development 2015-12-24 15:27:43 +01:00
Sebastian Dröge
7fddeaa878 pbutils: Link to libgstbase for bytewriter and adapter 2015-12-24 13:59:52 +01:00
Sebastian Dröge
5f98203bd7 Release 1.7.1 2015-12-24 13:59:15 +01:00
Sebastian Dröge
1975bdcd1c Update .po files 2015-12-24 13:10:08 +01:00
Sebastian Dröge
2361745117 po: Update translations 2015-12-24 12:22:04 +01:00
Thibault Saunier
512ac3ea72 encodebin: Implement an encoding profile serialization format
https://bugzilla.gnome.org/show_bug.cgi?id=759356
2015-12-24 09:52:53 +01:00
Koop Mast
d08b96b0b7 configure: Make -Bsymbolic check work with clang.
Update the -Bsymbolic check with the version glib has. This version
works with clang.

https://bugzilla.gnome.org/show_bug.cgi?id=759713
2015-12-21 12:27:58 +01:00
Kazunori Kobayashi
d43f1b2a5a appsrc: Clear is_eos flag when receiving the flush-stop event
The EOS event can be propagated to the downstream elements when
is_eos flag remains set even after leaving the flushing state.
This fix allows this element to normally restart the streaming
after receiving the flush event by clearing the is_eos flag.

https://bugzilla.gnome.org/show_bug.cgi?id=759110
2015-12-19 11:35:39 +01:00
Thiago Santos
8b05f682b0 examples: playback-test: remove unused variables
audiosink and videosink string variables are unused
2015-12-18 19:01:09 -03:00
Matthew Waters
023af2d3b1 playbin: only add the template caps when the result is empty
Unconditionally adding the template caps when proxying the caps query will play
havoc with decoders that attempt to choose an output format based on some caps
features.  Creating a sink that does not include those caps features and a
decoder/parser/etc that preferentially chooses some specific caps feature when
available, will always return the decoder/parser/etc template caps and choose a
feature that downstream will be unable to support.

Fix by limiting the addition of the template caps to when the result is actually
empty.

https://bugzilla.gnome.org/show_bug.cgi?id=758212
2015-12-18 21:55:00 +11:00
Sebastian Dröge
aadefefba8 configure: Don't use AG_GST_CHECK_FEATURE for checking for gio-unix-2.0
It's meant to be used for external plugins that can then all be disabled via
--disable-external. gio-unix-2.0 however is just an optional dependency for
the TCP unit test.

Also when using AG_GST_CHECK_FEATURE like this, in the --disable-external part
there needs to be an AM_CONDITIONAL for the feature with FALSE.
2015-12-17 13:39:01 +01:00
Sebastian Dröge
60bad4815d Revert "decodebin2: fix deadlock on chain shutdown"
This reverts commit 77dc09c3a9.

It can cause the FLUSH_START/STOP events to go to the sink elements, which
then causes state changes and various other problems. We shouldn't really
flush downstream here, the idea is to do *draining*.

Apart from that the testcase for the original bug here works without this
commit now.
2015-12-16 17:09:25 +01:00
Luis de Bethencourt
29cfb9a6d7 multifdsink: fix typo in GST_WARNING_OBJECT
This should make easier to parse the debug logs.
s/fnctl/fcntl
2015-12-16 11:12:03 +00:00
Vincent Penquerc'h
033ce9b20d videorate: remove dead code
Since the loops increasing count from 0 are always run at least
once (if count < 1), count will always be at least one when
compared to the drop/dup conditions.

Coverity 1139674
2015-12-16 11:00:22 +00:00
Wim Taymans
08734e7598 audio-converter: rework the main processing loop
Rework the main processing loop. We now create an audio processing
chain from small core functions. This is very similar to how the
video-converter core works and allows us to statically calculate an
optimal allocation strategy for all possible combinations of operations.
Make sure we support non-interleaved data everywhere.
Add functions to calculate in and out frames and latency.
2015-12-16 11:13:15 +01:00
Wim Taymans
8bcf183c7f audioconvert: clear convert object 2015-12-16 11:13:15 +01:00
Sebastian Dröge
559fd76d7d docs: update to git 2015-12-16 09:35:38 +01:00
Nicolas Dufresne
e7a59d2e08 Revert "alsasrc: Disable HW timestamp"
This reverts commit 3642e9a391.
2015-12-14 13:59:02 -05:00
Xavier Claessens
429860e51f base: Add g_autoptr() support to all types
https://bugzilla.gnome.org/show_bug.cgi?id=754464
2015-12-14 13:39:43 -05:00
Nicolas Dufresne
3642e9a391 alsasrc: Disable HW timestamp
This is a workaround for broken pulse module.
2015-12-14 13:39:02 -05:00
Sebastian Dröge
b0c834df1b rtspconnection: Properly initialize stack-allocated RTSP message to all-zeroes 2015-12-14 19:03:33 +01:00
Evan Callaway
5ac65d9e3a rtspconnection: Use relative URI for non-proxy tunneled requests
Match the section 5.1.2 of the HTTP/1.0 spec by using relative URIs unless we
are using a proxy server. Also, send Host header for compatability with
HTTP/1.1 and some HTTP/1.0 servers.

https://bugzilla.gnome.org/show_bug.cgi?id=758922
2015-12-14 18:21:10 +01:00
Evan Callaway
65c7bd7a2c rtspconnection: Support authentication during tunneling setup
gst_rtsp_connection_connect_with_response accepts a response pointer
which it fills with the response from setup_tunneling if the
connection is configured to be tunneled.  The motivation for this is to
allow the caller to inspect the response header to determine if
additional authentication is required so that the connection can be
retried with the appropriate authentication headers.

The function prototype of gst_rtsp_connection_connect has been
preserved for compatability with existing code and wraps
gst_rtsp_connection_connect_with_response.

https://bugzilla.gnome.org/show_bug.cgi?id=749596
2015-12-14 16:00:45 +01:00
Sebastian Dröge
d6be67265f rtpbasedepayload: Check if the packet loss event actually has timestamp and duration fields
CID 1139615
2015-12-14 13:11:21 +01:00
Wim Taymans
f5a3f70571 audio: adapt API for non-interleaved formats
Allow an array of sample blocks to be passed to the channel mix and
quantizer functions to support non-interleaved formats.
2015-12-14 09:16:08 +01:00
Wim Taymans
aec17c63fd audio-converter: improve API for non-interleaved formats
Make it possible to pass an array of sample blocks when dealing with
non-interleaved formats.
2015-12-14 09:16:08 +01:00
Luis de Bethencourt
055ed65d92 riff: add FourCC aliases
Support media using the aliases defined in http://www.fourcc.org/ that are
exact duplicates of already known codes.
2015-12-12 20:22:44 +00:00
Luis de Bethencourt
98e93ec5ee riff: use defined FourCC
Make gst_riff_create_video_caps() use the FourCC available in riff-ids.h,
like gst_riff_create_audio_caps() does.
2015-12-12 20:22:09 +00:00