Commit graph

684 commits

Author SHA1 Message Date
Francisco Velazquez
b8a97007cc docs: Fix broken URL in reference manual
https://bugzilla.gnome.org/show_bug.cgi?id=780566
2017-03-27 12:54:02 +01:00
Nicolas Dufresne
fb7d9e26ff Fix plugin filenames to match pugin names
- libgstencodebin.so is now libgstencoding.so
 - libgstximage.so is now libgstximagesink.so (meson only)

https://bugzilla.gnome.org/show_bug.cgi?id=779344
2017-03-08 20:04:17 -05:00
Sebastian Dröge
38ec8f396f rawparse: Move to gst-plugins-base
https://bugzilla.gnome.org/show_bug.cgi?id=774544
2017-02-25 14:48:40 +02:00
Sebastian Dröge
b078f0ad40 Release 1.11.2 2017-02-24 15:07:06 +02:00
Georg Lippitsch
b3df5786a9 videotimecode: Init from GDateTime
Add a function to init the time code from a GDateTime

https://bugzilla.gnome.org/show_bug.cgi?id=778702
2017-02-23 19:50:39 +02:00
Víctor Manuel Jáquez Leal
4fa6a2aba1 docs: update libs section
Include documented symbols that were not declared in section file.
2017-01-21 18:06:11 +01:00
Sebastian Dröge
b728e91ebb Release 1.11.1 2017-01-12 15:30:02 +02:00
Thibault Saunier
46b424a38b encoding-profile: Add a way to copy an encoding profile
It is often usefull to make sure that you get a full copy of a profile.
For example you want to let the user modify it in the user interface
but still keep an unchanged version for later use.

API:
  gst_encoding_profile_copy
2017-01-06 11:40:20 -03:00
Evan Nemerson
98064ed9bf audioringbuffer: add set_callback_full() for g-i
https://bugzilla.gnome.org/show_bug.cgi?id=678301
2016-12-22 15:34:58 +00:00
Thibault Saunier
6f50b59a20 meson:doc: Build libraries documentations 2016-12-16 11:27:31 -03:00
Tim-Philipp Müller
6bb574ba89 docs: design: remove outdated draft docs (hw-acceleration, va) 2016-12-08 23:15:31 +00:00
Tim-Philipp Müller
46138b1b1d docs: design: move most design docs to gst-docs module 2016-12-08 23:00:45 +00:00
Sebastian Dröge
90b24d34b3 rtsp: Add gst_rtsp_message_parse_auth_credentials() to parse authentication credentials
https://bugzilla.gnome.org/show_bug.cgi?id=774416
2016-11-21 09:39:21 +02:00
Sebastian Dröge
828c8604dd rtsp: Add gst_rtsp_generate_digest_auth_response() to calculate digest auth response
https://bugzilla.gnome.org/show_bug.cgi?id=774416
2016-11-21 09:39:21 +02:00
Julien Isorce
3bf893e12a video: add gst_video_decoder_allocate_output_frame_with_params
It adds a third argument to pass GstBufferPoolAcquireParams
to gst_buffer_pool_acquire_buffer.

If a user subclasses GstBufferPoolAcquireParams, this allows to
pass an updated param to the underlying buffer pool at each
gst_video_decoder_allocate_output_frame_with_params call.

https://bugzilla.gnome.org/show_bug.cgi?id=773165
2016-11-04 16:18:13 +00:00
Julien Isorce
f5eb366335 allocators: define GST_CAPS_FEATURE_MEMORY_DMABUF
Adds "memory:DMABuf" caps feature. Since 1.11 tag.
Useful when the the dma-buf buffer cannot be mapped to CPU for r/w requests.
Example: protected content or platform constraints.

https://bugzilla.gnome.org/show_bug.cgi?id=759358
2016-11-03 13:19:12 -04:00
Nicolas Dufresne
c37b1e8c56 dmabuf: Make the allocator sub-classable
This should allos for cleaner code when implement such allocator.

https://bugzilla.gnome.org/show_bug.cgi?id=768794
2016-11-03 13:19:12 -04:00
Sebastian Dröge
79809633de video-info: Add optional field-order caps field for interlaced-mode=interleaved
Usually this information is static for the whole stream, and various
container formats store this information inside the headers for the
whole stream.

Having it inside the caps for these cases simplifies code and makes it
possible to express these requirements more explicitly with the caps.

https://bugzilla.gnome.org/show_bug.cgi?id=771376
2016-11-01 20:40:07 +02:00
Sebastian Dröge
9c3043470e Release 1.10.0 2016-11-01 17:53:24 +02:00
Sebastian Dröge
45a04f9d8b Release 1.9.90 2016-09-30 13:01:53 +03:00
Sebastian Dröge
47b7c8dc75 Release 1.9.2 2016-09-01 12:26:20 +03:00
Xabier Rodriguez Calvar
0341f04ce1 videodirection: interface for rotation and flip
A GstVideoOrientationMethod enumeration is also provided for the
admitted property values.

https://bugzilla.gnome.org/show_bug.cgi?id=768687
2016-08-25 10:19:13 +03:00
Sebastian Dröge
7f7d667e0f videotimecode: Add to docs and exports list 2016-08-04 19:06:45 +03:00
Tim-Philipp Müller
5044bf79ae Add more files to .gitignore 2016-07-23 14:42:30 +01:00
Tim-Philipp Müller
f25eee6f4b docs: add playbin3, decodebin3, parsebin, urisourcebin to docs
Docs still need some fleshing out though.
2016-07-22 12:52:12 +01:00
Joan Pau Beltran
c6722c06a0 appsink: add _pull_sample/preroll() variants with timeout
The _pull_sample() and _pull_preroll() functions block
until a sample is available, EOS happens or the pipeline
is shut down (returning NULL in the last two cases).

This adds _try_pull_sample() and _try_pull_preroll()
functions with a timeout argument to specify the maximum
amount of time to wait for a new sample.

To avoid code duplication, wait forever if the timeout is
GST_CLOCK_TIME_NONE and use that to implement
_pull_sample/_pull_preroll with the original behavior.

Add also corresponding action signals "try-pull-sample"
and "try-pull-preroll".

https://bugzilla.gnome.org/show_bug.cgi?id=768852
2016-07-18 16:55:16 +01:00
Sebastian Dröge
08f993d090 Release 1.9.1 2016-07-06 13:06:06 +03:00
Nicolas Dufresne
950c73af43 plugin-doc: Minor re-order 2016-06-21 14:54:10 -04:00
Nicolas Dufresne
6123791399 Automatic update of plugins doc files 2016-06-21 14:54:10 -04:00
Joan Pau Beltran
bd49854c32 docs: design: add IYU2 raw video format description
https://bugzilla.gnome.org/show_bug.cgi?id=763026
2016-06-01 15:09:42 +01:00
Sebastian Dröge
cebddd5103 docs: Update for git master 2016-05-15 13:31:03 +03:00
Sebastian Dröge
dc8120f298 appsrc: Add duration property for providing a duration in TIME format
https://bugzilla.gnome.org/show_bug.cgi?id=766229
2016-05-10 16:50:32 +03:00
Guillaume Desmottes
3cb08304da gst-audio: add gst_audio_channel_positions_to_string()
We currently don't log much about channel positions making debugging
harder as it should be. This is the first step in my attempt to improve
this.

https://bugzilla.gnome.org/show_bug.cgi?id=763985
2016-04-12 14:48:30 -04:00
Jimmy Ohn
65f721b326 codec-utils: Add utilities for AAC and the AACHead header
Add utilities about the channels and sample rate for AAC.

https://bugzilla.gnome.org/show_bug.cgi?id=749110
2016-03-24 14:27:21 +02:00
Sebastian Dröge
d67525d594 Release 1.8.0 2016-03-24 12:19:23 +02:00
Sebastian Dröge
a730be9cbd Release 1.7.91 2016-03-15 12:02:20 +02:00
Sebastian Dröge
48f584e663 Release 1.7.90 2016-03-01 18:14:54 +02:00
Tim-Philipp Müller
fd9e22e5e9 docs: add Opus to docs 2016-02-26 00:37:57 +00:00
Sebastian Dröge
97e108beba Release 1.7.2 2016-02-19 11:48:30 +02:00
Tim-Philipp Müller
56be0653e0 docs: fix up for GstAudioChannelMix rename as well 2016-01-08 16:37:25 +00:00
Tim-Philipp Müller
63d7a2a89a docs: add new audio API 2016-01-05 22:52:34 +00:00
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
5f98203bd7 Release 1.7.1 2015-12-24 13:59:15 +01:00
Sebastian Dröge
559fd76d7d docs: update to git 2015-12-16 09:35:38 +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
Matthew Waters
0b98ed32ce videometa: add GstVideoAffineTransformationMeta
Adds a simple 4x4 affine transformations meta for passing arbitrary
transformations on buffers.

Based on patch by Matthieu Bouron

https://bugzilla.gnome.org/show_bug.cgi?id=731791
2015-11-11 00:19:25 +11:00
Wim Taymans
e3f0f3b91e audio-converter: move audio converter to audio libs
Move the audio-converter helper to the audio library.
2015-11-06 17:53:22 +01:00
Sebastian Dröge
bcd7b2fff2 codec-utils: Add utilities for Opus caps and the OpusHead header
https://bugzilla.gnome.org/show_bug.cgi?id=757152
2015-11-03 20:35:33 +02:00
Sebastian Dröge
35ea6fdddf audio: Add GstAudioClippingMeta for specifying clipping on encoded audio buffers
https://bugzilla.gnome.org/show_bug.cgi?id=757153
2015-11-03 20:35:33 +02:00
Stefan Sauer
caf7b6674b docs: add alsamidisrc to docs 2015-10-01 21:53:20 +02:00