mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-22 00:06:36 +00:00
Code Issues 1 Projects Releases Wiki Activity
14712 commits 44 branches 1510 tags 505 MiB
Commit graph

4533 commits

Author SHA1 Message Date
Sebastian Dröge
2f3eb47a95 audiobasesrc: Post latency message on the bus after set_caps() 
The latency is only known once the caps are known, and might change
whenever the caps are changing.

https://bugzilla.gnome.org/show_bug.cgi?id=758911
 2015-12-01 19:58:25 +02:00
Michael Olbrich
43155807cd audiobasesink: Post latency message on the bus after set_caps() 
Any latency query before this will not get the correct latency so a new
latency query should be triggered once the audio sink know its own latency.

Without this the initial latency query from the pipeline arrives too early
sometimes and the resulting latency is too short.

https://bugzilla.gnome.org/show_bug.cgi?id=758911
 2015-12-01 19:58:25 +02:00
Luis de Bethencourt
2a70c86e85 codec-utils: accept wrong version field in OpusHead header 
Some Opus files found on the wild have 0 in the version field of the
OpusHead header, instead of the correct value of 1. The files still
play, don't make this error fatal.

https://bugzilla.gnome.org/show_bug.cgi?id=758754
 2015-12-01 15:47:35 +00:00
William Manley
aae0dc37c9 allocators: add debug category for fd memory and allocator 
Debugging can now be viewed by setting GST_DEBUG=fdmemory:9

https://bugzilla.gnome.org/show_bug.cgi?id=758744
 2015-11-27 15:33:47 +00:00
Ravi Kiran K N
df5725e683 id3v2frames: Handle private frames 
Handle PRIV ID3 tag having owner information (string)
and binary data, add to tag messages list.

https://bugzilla.gnome.org/show_bug.cgi?id=730926
 2015-11-20 20:20:18 +00:00
Tim-Philipp Müller
93a92d7f70 tags: id3: make sure to register private-id3v2-frame tag before using it 2015-11-20 19:15:22 +00:00
Ognyan Tonchev
7a702df863 rtspconnection: Add support for parsing custom headers 
https://bugzilla.gnome.org/show_bug.cgi?id=758235
 2015-11-18 00:15:32 +00:00
Reynaldo H. Verdejo Pinochet
0c95b0a738 Remove unnecessary NULL checks before g_free() 
g_free() is NULL-safe
 2015-11-17 14:50:27 -08:00
Luis de Bethencourt
09c881ee14 codec-utils: guint8 can't hold value over 255 
channels is a guint8, so the max value is 255 and checking if it value is
> 256 will never be false.

CID 1338687, CID 1338688
 2015-11-12 14:39:22 +00:00
Luis de Bethencourt
df16e8dd5a audio-converter: remove unneeded check for unsigned < 0 
Commit ff6d1a2a25 changed sample's type from
gint to gsize (and renamed it to in_samples). gsize is an unsigned long,
which means it can never be a negative value and the check making sure that
in_samples is >= 0 is never going to be false. Removing it.

CID 1338689
 2015-11-12 14:18:30 +00:00
Vineeth TM
3f099e3c29 pbutils:encoding-target: Fix string memory leak 
https://bugzilla.gnome.org/show_bug.cgi?id=757926
 2015-11-11 15:40:52 +01:00
Vineeth TM
b61e1465b7 audio-quantize: Fix dither_buffer memory leak 
https://bugzilla.gnome.org/show_bug.cgi?id=757928
 2015-11-11 15:01:08 +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
ff6d1a2a25 audio-converter: add output size argument 
Make it possible to have a different number of output samples than input
samples when we, for example, want to add resampling later.
 2015-11-10 09:53:59 +01:00
Thibault Saunier
629b63d1f2 discoverer: Check API arguments and assert if needed 2015-11-07 00:46:47 +01:00
Wim Taymans
30977cf1a5 audio-converter: require interleaved samples and no resampling 
We can't yet do resampling or anything other than interleaved audio.
 2015-11-06 18:00:41 +01:00
Wim Taymans
7abed02858 audio: update ORC dist files 2015-11-06 17:54:21 +01:00
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
Wim Taymans
dfa25a40fc audio-channel-mix: move channel mixer to audio libs 
Move the channel mixer code to the audio library
 2015-11-06 17:39:33 +01:00
Wim Taymans
b8bea9d8be audio: add debug categories 2015-11-06 17:29:22 +01:00
Wim Taymans
59db8ce542 audio-quantize: update docs 
Update docs
Add another flag for the quantizer
 2015-11-06 13:02:19 +01:00
Wim Taymans
dfbeb78342 audio: update orc files 2015-11-06 12:37:14 +01:00
Wim Taymans
c36ac3ce45 audioconvert: move audio quantize code to libs 
Move the audio quantize code from audioconvert to the audio library.
work on making an audio converter helper function similar to the video
converter.
Fold fastrandom directly into the quantizer, add some ORC code to
optimize this later.
 2015-11-06 12:10:48 +01:00
Wim Taymans
a7789854d5 audio-channels: rename get_default_mask 
Rename _get_default_mask() to _get_fallback_mask() to make it more
clear that the function only provides a fallback if nothing else can be
done. Also clarify this in the documentation.

API: gst_audio_channel_get_fallback_mask()
 2015-11-05 12:50:18 +01:00
Wim Taymans
f86ed8cdf6 audio-channels: make method to get default channel-mask 
Add a new method to get the default channel-mask.
Use the new method on audiodecoder and audioconvert.

API: gst_audio_channel_get_default_mask()
 2015-11-05 10:52:53 +01:00
Arnaud Vrac
dfe250d17d video: blend using OVER operation 
Also support all premultiplied/non-premultiplied source/destination
configurations

https://bugzilla.gnome.org/show_bug.cgi?id=681447
 2015-11-04 21:58:32 +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
Tim-Philipp Müller
1f2fdd3789 audio: update disted orc backup files 2015-11-03 16:38:09 +00:00
Luis de Bethencourt
94a7f9fc4e audioclock: use GST_STIME_FORMAT for GstClockTimeDiff 
GST_STIME_FORMAT is more appropriate for GstClockTimeDiff since it can
handle negative values better.

https://bugzilla.gnome.org/show_bug.cgi?id=757480
 2015-11-03 14:08:29 +00:00
Luis de Bethencourt
227f1d1e0f videodecoder: Print GstClockTimeDiff as a signed integer in debug logs 2015-11-03 13:44:39 +00:00
Wim Taymans
801f7ca464 audio-format: add TRUNCATE_RANGE flag 
Add a TRUNCATE_RANGE flag for unpack functions to fill the least
significate bits with 0 (as did the old code). Also add functions
that don't truncate. Use the TRUNC flag in audioconvert for
backwards compatibility for now.
 2015-11-03 12:12:08 +01:00
Wim Taymans
914aa4aed1 audiopack: improve pack functions 
Avoid shifts by using convh functions.
 2015-11-03 12:12:08 +01:00
Luis de Bethencourt
fe62e797d5 audiobasesink: use GST_STIME_ARGS for GstClockTimeDiff 
No need to use G_GINT64_FORMAT for potentially negative values of
GstClockTimeDiff. Since 1.6 these can be handled with GST_STIME_ARGS.
Plus it creates more readable values in the logs.

https://bugzilla.gnome.org/show_bug.cgi?id=757480
 2015-11-02 17:35:20 +00:00
Sebastian Dröge
443171bb4c audio: Fix parameters to gst_buffer_get_audio_downmix_meta() in macro 2015-11-02 17:35:45 +02:00
Luis de Bethencourt
b81b3f07ec videodecoder: subtract time difference with GST_CLOCK_DIFF 
To ensure the subtraction of two GstClockTime values (which are guint64)
can be negative. Use GST_CLOCK_DIFF which returns a gint64.

CID 1338049
 2015-11-02 12:09:45 +00:00
Thibault Saunier
a7123ebb58 encoding-profile: Do not force user to provide an encoding profile name 
And use the profile called `default` if none provided.
 2015-11-02 11:35:55 +01:00
Thibault Saunier
83fa06aab5 encoding-target: Do not unconditionally break when searching for a target 
Otherwise the loop is useless!

Fixes CID 1338051
 2015-11-02 11:31:34 +01:00
Sebastian Dröge
736a27fe1e audiofilter: Clip input buffers to the segment before handling them 
https://bugzilla.gnome.org/show_bug.cgi?id=757068
 2015-11-02 10:20:37 +02:00
Sebastian Dröge
1da79c76a7 videodecoder: Print another time difference as a signed integer instead of a huge unsigned one 2015-10-29 16:52:49 +02:00
Sebastian Dröge
f17758d9e3 videodecoder: Print GstClockTimeDiff as a signed integer in debug logs 2015-10-29 16:01:26 +02:00
Thibault Saunier
2e20f3ba4f encoding-target: Add a GST_ENCODING_TARGET_PATH envvar to find target files 2015-10-28 16:07:44 +01:00
Thibault Saunier
96f69fa998 encoding-target: Allow having encoding target without a category set 
There was already some code to handle that, but the support was not
complete in those code paths.
 2015-10-28 16:07:44 +01:00
Thibault Saunier
0256381f6f encoding-target: Create directory before trying to save encoding targets 2015-10-28 16:07:44 +01:00
Thibault Saunier
db272cf9cb encoding-profile: Allow specifying the target category in the serialized encoding target 2015-10-28 16:07:44 +01:00
Pavel Bludov
8051434c03 video: overlay-composition: fix rectangle and composition cast macros 
Closing parenthesis was missing in two cases.

https://bugzilla.gnome.org/show_bug.cgi?id=756893
 2015-10-22 09:24:01 +01:00
Sebastian Dröge
5bdeed32bc rtp: GST_RTP_BUFFER_MAP_FLAG_SKIP_PADDING is Since 1.6.1 2015-10-20 12:10:52 +03:00
eunhae choi
e98b96247f audiobasesink: fix issue about eos handling during flushing 
If the flush-start is arrived during _eos_wait() in basesink,
the 'eos' flag is overwritten to TRUE after exiting the _eos_wait().
To resolve the overwritten issue,
the subclass doing the _eos_wait() call should return the right value.
If the eos flag is set to TRUE again, it will cause error(enter the eos flow)
of the following state changing from PAUSED to PLAYING in basesink.

https://bugzilla.gnome.org/show_bug.cgi?id=754980
 2015-10-19 12:12:12 -03:00
Sebastian Dröge
df0d5aa4d5 video: Add out annotations to the out parameters of gst_video_calculate_display_ratio() 
https://bugzilla.gnome.org/show_bug.cgi?id=754567
 2015-10-17 20:37:22 +03:00
Stian Selnes
0a668c1866 rtpbuffer: Add map flag to skip padding 
Encrypted RTP buffers may contain encrypted padding, hence it's
necessary to have an option to relax the validation in order to
successfully map the buffer.

When the flag GST_RTP_BUFFER_MAP_FLAG_SKIP_PADDING is set
gst_rtp_buffer_map() will map the buffer like if padding is not
present.

https://bugzilla.gnome.org/show_bug.cgi?id=752705
 2015-10-15 22:42:37 +03:00