Commit graph

2850 commits

Author SHA1 Message Date
Wim Taymans
8d93f8edb3 videopool: add support for custom allocators 2011-12-28 15:35:52 +01:00
Tim-Philipp Müller
3dfdd6be9d audioringbuffer: rename GST_BUFTYPE_* to GST_AUDIO_RING_BUFFER_FORMAT_TYPE_*
Bit unwieldy, but more appropriate. Could also be moved into
audio.h as GstAudioFormatType.
2011-12-25 21:38:21 +00:00
Tim-Philipp Müller
80095caa40 audioringbuffer: remove unused GstAudioRingBufferSegState enum and field 2011-12-25 21:23:11 +00:00
Tim-Philipp Müller
dfa23662e1 tag: fix writing of Exif tag payloads <= 4 bytes
When the payload for an Exif tag is less than or equal to 4 bytes,
the data is simply put into the offset field. Fix writing these
kinds of payloads on big endian systems (and possibly also on
little endian systems). The caller will have already formatted
the bytes in memory according to the writer's endianness, so just
write out the bytes as they are in this case. Fixes tags unit test
on big endian systems.
2011-12-23 22:24:44 +00:00
Mark Nauwelaerts
e3c78ff661 audioencoder: add a few more debug statements 2011-12-22 16:58:37 +01:00
Mark Nauwelaerts
9bfa65b7d3 audiodecoder: tweak documentation 2011-12-22 16:58:34 +01:00
Thiago Santos
77353dad21 tag: xmp: Keep compatibility with our old generated xmp
We used to add a trailing \n to the end of generated xmp packets.
Windows viewer was unhappy with it and we fixed it in
96d2120c2b

The problem is that this caused xmp generated before this fix
to not be recognized and parsed anymore. This patch makes it
recognize xmp with the trailing \n and without, fixing the
regression. Also adds tests for it.
2011-12-22 08:12:28 -03:00
Wim Taymans
a5f3d21723 videofilter: improve video filter
Flesh out the video filter base class. Make it parse the input and output caps
and turn them into GstVideoInfo. Map buffers as video frames and pass them to
the transform functions.
This allows us to also implement the propose and decide_allocation vmethods.
Implement the transform size method as well.
Update subclasses with the new improvements.
2011-12-21 23:46:53 +01:00
Wim Taymans
f50573fd66 videopool: add videopool implementation
Add a GstVideoPool object that can be used to allocate video frames with support
for metadata and alignment.
Add method to reset alignment info.
2011-12-21 18:13:17 +01:00
Wim Taymans
ddc05e0ed1 propertyprobe: remove propertyprobe
Remove the propertyprobe interface
Improve docs
2011-12-21 11:58:53 +01:00
Vincent Penquerc'h
ceeff69bc1 gstvideo: fix a RGB ordering mixup in colorspace conversion code 2011-12-20 19:55:53 +00:00
Sebastian Dröge
2760dd2068 audiobasesrc: Use guint8 instead of guchar 2011-12-20 14:36:28 +01:00
Sebastian Dröge
338622fe7e audioringbuffer: Use guint8 instead of guchar 2011-12-20 14:36:28 +01:00
Mark Nauwelaerts
c41f3cbef0 audiodecoder: set a non-zero default maximum tolerated errors
Whereas the previous default 0 was backwards compatible in that it lead
to erroring out immediately upon any error, elements that are really
ported and using the base class error macro can be assumed to intend to
improve behaviour rather than maintaining the old one.  So, make it easy
on those and any future one and tolerate some errors by default, as intended.

Fixes #666579.
2011-12-20 12:50:18 +01:00
Wim Taymans
7505b7a55c add audio metadata
Add some audio metadata to describe a downmix matrix.
Add metadata to media type document.
2011-12-20 12:02:25 +01:00
Wim Taymans
6241ff63f8 video: update interlace caps and docs
Remove interlaced boolean from caps and replace with an interlace-mode enum.
document this new property in the video caps document. With the enum we can
put fields into separate video meta.
Add enum for this interlace-mode in the VideoInfo.
Update the buffer flags.
2011-12-19 18:03:45 +01:00
Wim Taymans
81d351b5f0 tagdemux: add FIXME
Add a FIXME because the EOS before-type case now has to be solved differently
because the srcpad is always available.
2011-12-19 11:03:55 +01:00
Thiago Santos
d798cc1b8d tag: exif: do not include \0 in size passed to g_convert
When using g_convert, we should only pass the length
of the string content (without the \0) as g_convert will
only parse the real contents when changing formats. Including
the \0 causes it to add another \0, increasing the string
size when not needed.

For example, when writting a North geo location ref entry, that should
be a string with a single N letter, it would write:
"N\0\0", causing the string to have size 3, instead of 2 as expected.

In our case, we can pass -1 and let g_convert calculate the strlen as
we don't use the length anywhere else.

This fixes jifmux's tests on gst-plugins-bad.
2011-12-15 12:08:51 -03:00
Christian Fredrik Kalager Schaller
0d552ae53d Fix 666168, add missing allow-None to encodebin APIs 2011-12-14 17:34:55 +00:00
Tim-Philipp Müller
d5ae24fe91 encoding-profile: add some missing allow-none g-i annotations
Fix gst_encoding_container_profile_new() annotations.

https://bugzilla.gnome.org/show_bug.cgi?id=666096
2011-12-14 12:28:26 +00:00
Stefan Sauer
d0a5cb8c01 riff-media: port GST_BUFFER_DATA to 0.11 in conditional code branch 2011-12-14 11:31:31 +01:00
Vincent Penquerc'h
12be1e6fc5 baseaudiosink: fix late buffer leak 2011-12-13 12:55:45 +00:00
Sebastian Dröge
739de5fbf9 glib-compat: Add license boilerplate for LGPL 2011-12-13 00:03:28 +00:00
Wim Taymans
59d5ad42b0 rtsp: use rtpbin 2011-12-09 19:22:21 +01:00
Wim Taymans
6be9a67148 rtp: add INIT macros 2011-12-09 19:22:21 +01:00
Tim-Philipp Müller
54c5cd8c3f rtpbuffer: add GST_RTP_BUFFER_INIT to initialize RTP buffers on the stack
Fixes build of -good.
2011-12-09 15:03:41 +00:00
Tim-Philipp Müller
fb6d09055a Merge remote-tracking branch 'origin/master' into 0.11
Conflicts:
	ext/alsa/gstalsadeviceprobe.c
	ext/alsa/gstalsamixer.c
	ext/pango/gsttextoverlay.c
	ext/pango/gsttextoverlay.h
	gst-libs/gst/audio/gstaudiobasesink.c
	gst-libs/gst/audio/gstaudioringbuffer.c
	gst-libs/gst/audio/gstaudiosrc.c
	gst-libs/gst/video/Makefile.am
	gst-libs/gst/video/video.c
	gst/encoding/gststreamcombiner.c
	gst/encoding/gststreamsplitter.c
	gst/playback/gstplaybasebin.c
	gst/playback/gststreamsynchronizer.c
	gst/playback/gstsubtitleoverlay.c
	gst/playback/gsturidecodebin.c
	sys/xvimage/xvimagesink.c
	tests/examples/Makefile.am
	win32/common/libgstvideo.def

Video overlay composition disabled for now, needs
porting to buffer meta.
2011-12-08 01:19:03 +00:00
Tim-Philipp Müller
91bbfbd819 video: make composition_blend() return a boolean
Not that anyone will ever check that, and it's not clear what
they're supposed to do if it fails, but at least it's there.
2011-12-07 18:45:28 +00:00
Tim-Philipp Müller
6757afc0bc docs: add new API to docs 2011-12-07 18:38:06 +00:00
Tim-Philipp Müller
5037b39883 video: add seqnum getters for overlay compositions and rectangles
API: gst_video_overlay_composition_get_seqnum()
API: gst_video_overlay_rectangle_get_seqnum()
2011-12-07 17:57:08 +00:00
Thibault Saunier
39c5015ed0 video: support any type of video in _parse_caps
Slight change in semantics for convenience. Shouldn't cause any
problems since this function is usually only used on pre-filtered
caps and not random caps, and it's hard to imagine a situation
where someone would want to rely on the previous behaviour.
2011-12-07 15:51:14 +00:00
Tim-Philipp Müller
61d0ab1faa video: fix leak in gst_video_format_new_template_caps()
g_value_reset() is not the same as g_value_unset()
2011-12-06 14:55:54 +00:00
Wim Taymans
f096b8a8d8 ringbuffer: remove old _full version 2011-12-06 15:06:12 +01:00
Wim Taymans
9e97260c9f fix for basesrc changes 2011-12-06 13:59:11 +01:00
Edward Hervey
ea0ed511f8 rtp: Initialize GstRTPBuffer before usage 2011-12-05 18:42:24 +01:00
Edward Hervey
94230af7a3 rtp: Don't forget to initialize GstRTPBuffer 2011-12-05 18:30:37 +01:00
Tim-Philipp Müller
7d20a7bdb9 video: don't use deprecated GStaticMutex with newer glib versions 2011-12-05 15:48:07 +00:00
Tim-Philipp Müller
6630236af4 video: add video overlay composition API for subtitles
Basic API to attach overlay rectangles to buffers,
or blend them directly onto raw video buffers.

To be used primarily for things like subtitles or
logo overlays, not meant to replace videomixer.

Allows us to associate subtitle overlays with
non-raw video surface buffers, so that subtitles
are not lost and can instead be rendered later
when those surfaces are displayed or converted,
whilst re-using all the existing overlay plugins
and not having to teach them about our special
video surfaces. Could also have been made part
of the surface buffer abstraction of course, but
a secondary goal was to consolidate the blending
code for raw video into libgstvideo, and this
kind of API allows us to do both in a way that's
minimally invasive to existing elements, and at
the same time is fairly intuitive.

More features and extensions like the ability to
pass the source data or text/markup directly will
be added later.

https://bugzilla.gnome.org/show_bug.cgi?id=665080

API: gst_video_buffer_get_overlay_composition()
API: gst_video_buffer_set_overlay_composition()

API: gst_video_overlay_composition_new()
API: gst_video_overlay_composition_add_rectangle()
API: gst_video_overlay_composition_n_rectangles()
API: gst_video_overlay_composition_get_rectangle()
API: gst_video_overlay_composition_make_writable()
API: gst_video_overlay_composition_copy()
API: gst_video_overlay_composition_ref()
API: gst_video_overlay_composition_unref()

API: gst_video_overlay_composition_blend()

API: gst_video_overlay_rectangle_new_argb()
API: gst_video_overlay_rectangle_get_pixels_argb()
API: gst_video_overlay_rectangle_get_pixels_unscaled_argb()
API: gst_video_overlay_rectangle_get_render_rectangle()
API: gst_video_overlay_rectangle_set_render_rectangle()
API: gst_video_overlay_rectangle_copy()
API: gst_video_overlay_rectangle_ref()
API: gst_video_overlay_rectangle_unref()
2011-12-05 15:37:02 +00:00
Tim-Philipp Müller
a7a3f969b3 video: hide private video-blend.[ch] from gobject-introspection
And remove unused fields from helper structure.
2011-12-05 15:37:02 +00:00
Tim-Philipp Müller
b0ff1d22e9 video: add fallbacks for compilation without orc 2011-12-05 15:36:56 +00:00
Thibault Saunier
80054a3b1e video: add some internal helper functions for image blending
This could be improved if we decide we don't need it to
be this generic/flexible.
2011-12-05 15:03:47 +00:00
Sebastian Dröge
b0f4085f22 xoverlay: Fix mistakes in the sample code
Fixes bug #665430.
2011-12-05 09:39:08 +01:00
Matej Knopp
f89d7ee7eb Appsink fixes 2011-12-05 09:34:50 +01:00
Tim-Philipp Müller
5440ae3c18 Suppress deprecation warnings in selected files, for g_static_rec_mutex_* mostly
GStaticRecMutex is part of our API/ABI, not much we can do here
in 0.10 for most of these.
2011-12-04 20:50:25 +00:00
Tim-Philipp Müller
0d98aa25b8 Work around deprecated thread API in glib master
Add private replacements for deprecated functions such as
g_mutex_new(), g_mutex_free(), g_cond_new() etc., mostly
to avoid the deprecation warnings. We'll change these
over to the new API once we depend on glib >= 2.32.

Replace g_thread_create() with g_thread_try_new().
2011-12-04 17:16:30 +00:00
Tim-Philipp Müller
6098442bd0 xmpwriter: update for thread API deprecations in glib master 2011-12-04 15:23:21 +00:00
Wim Taymans
1225aa9a78 update for basesink event handler changes 2011-12-02 22:24:43 +01:00
Tim-Philipp Müller
177525f89f Merge remote-tracking branch 'origin/master' into 0.11
Conflicts:
	gst-libs/gst/netbuffer/gstnetbuffer.c
	gst/ffmpegcolorspace/avcodec.h
	gst/ffmpegcolorspace/gstffmpegcodecmap.c
	gst/ffmpegcolorspace/imgconvert.c
	gst/ffmpegcolorspace/imgconvert_template.h
	gst/ffmpegcolorspace/mem.c
	gst/playback/README
	gst/playback/gstplaybasebin.c
	gst/playback/gstplaybasebin.h
	gst/playback/gstplaybin.c
	sys/v4l/v4lmjpegsrc_calls.c
	sys/v4l/videodev_mjpeg.h
	tests/check/elements/gnomevfssink.c
2011-12-02 11:10:17 +00:00
Piotr Fusik
14644457b0 various: typo fixes
Fix typos in code and docs. Fixes. #658984
2011-12-02 12:03:27 +01:00
Tim-Philipp Müller
ec0d3566bf Merge remote-tracking branch 'origin/master' into 0.11
Conflicts:
	ext/alsa/gstalsasrc.c
	ext/alsa/gstalsasrc.h
	gst/adder/gstadder.c
	gst/playback/gstplaybin2.c
	gst/playback/gstplaysinkconvertbin.c
	win32/common/libgstvideo.def
2011-12-02 00:07:39 +00:00
Wim Taymans
3deaa582d9 tags: make the tag functions return GstSample
gst_tag_image_data_to_image_buffer() ->
   gst_tag_image_data_to_image_sample() And make it return a GstSample.
Store the image-type into the extra sample info.
Remove a deprecated tag
2011-12-01 18:51:51 +01:00
Wim Taymans
59113af604 Use the new GstSample for snapshots
Make appsink return a GstSample. Remove the pull_buffer_list method because it
is not very useful anymore.
Pass GstSample to the conversion function.
Update playbin2 and examples
2011-12-01 16:53:11 +01:00
Wim Taymans
66d7151787 update marshal list 2011-12-01 15:54:49 +01:00
Wim Taymans
92ac25bdb3 videometa: add copy functions
Without copy functions, the metadata is lost when we make a buffer copy such as
when we make a buffer writable.
2011-12-01 15:45:28 +01:00
Wim Taymans
e064f9dbf6 appsrc: fix negotiation
Remove old useless caps code.
Make a negotiate function and use the configured caps as the caps on the appsrc
pad. If nothing was configured, fall back to the parent implementation.
2011-12-01 15:38:10 +01:00
Edward Hervey
e44db979f9 audio: Add audio-marshal.list to dist-ed files 2011-11-30 11:33:41 +01:00
Wim Taymans
47cbb230e9 audio: move audio interfaces
Move the audio related interfaces to the audio library.
2011-11-30 07:57:02 +01:00
Wim Taymans
4fb0f98bb9 encoding-profile: small cleanup in docs 2011-11-30 07:23:07 +01:00
Edward Hervey
5bc6ffcd8b video: Don't forget to install moved header files 2011-11-29 19:49:50 +01:00
Wim Taymans
871b306fce video: move some interfaces
Move some interfaces to the video library
2011-11-29 19:10:01 +01:00
Alessandro Decina
848711706b libgstvideo: minor fixes to key unit events
Make out args to gst_video_event_parse_{downstream|upstream}_force_key_unit
optional, update libgstvideo.def and fix docs a bit.

API: gst_video_event_new_upstream_force_key_unit
API: gst_video_event_new_downstream_force_key_unit
API: gst_video_event_is_force_key_unit
API: gst_video_event_parse_upstream_force_key_unit
API: gst_video_event_parse_downstream_force_key_unit

https://bugzilla.gnome.org/show_bug.cgi?id=607742
2011-11-29 09:15:59 +01:00
Andoni Morales Alastruey
df44e771f1 libgstvideo: Add force key unit events 2011-11-29 08:58:28 +01:00
Tim-Philipp Müller
0d87fd7146 Merge remote-tracking branch 'origin/master' into 0.11
Conflicts:
	gst-libs/gst/fft/gstffts16.h
2011-11-28 21:25:11 +00:00
Tim-Philipp Müller
0c056a04fe Merge commit '4a58223e4c824fedc024af435337a769e8ce593e' into 0.11 2011-11-28 21:20:10 +00:00
Philippe Normand
0a841f6712 fft: Bracket public headers
This is especially needed if the gstfftw library is used from C++
code.

Fixes #665074
2011-11-28 20:28:19 +01:00
Wim Taymans
5b868bd424 Update for indexable change 2011-11-28 18:24:03 +01:00
Wim Taymans
468d1dde89 audio: update for clock provider API change 2011-11-28 17:51:41 +01:00
Vincent Penquerc'h
ea78b060a7 Revert "libgstvideo: add a new API to handle QoS events and dropping logic"
This reverts commit eb03323fb6.

*grumble* I managed to merge something I did not mean to.
2011-11-28 13:26:53 +00:00
Vincent Penquerc'h
96374054ac various: fix pad template leaks
https://bugzilla.gnome.org/show_bug.cgi?id=662664
2011-11-28 13:09:02 +00:00
Vincent Penquerc'h
eb03323fb6 libgstvideo: add a new API to handle QoS events and dropping logic
https://bugzilla.gnome.org/show_bug.cgi?id=658241
2011-11-28 12:34:43 +00:00
Mark Nauwelaerts
4a58223e4c audioencoder: elaborate some documentation 2011-11-28 11:37:33 +01:00
Mark Nauwelaerts
9f57d91137 audiodecoder: add some documentation 2011-11-28 11:37:27 +01:00
Mark Nauwelaerts
856a5dd581 audiodecoder: really discard NULL decoded frame altogether
... including any timestamp, rather than having that one influence base_ts.
2011-11-28 11:37:23 +01:00
Tim-Philipp Müller
32b14c6ed3 Merge remote-tracking branch 'origin/master' into 0.11
Conflicts:
	ext/vorbis/gstvorbisenc.c
	gst/playback/gstdecodebin2.c
	gst/playback/gstplaysinkconvertbin.c
	gst/videorate/gstvideorate.c
2011-11-26 12:12:59 +00:00
Tim-Philipp Müller
a0639dad38 audio: remove unstable API guards from the audio decoder and encoder base classes 2011-11-25 13:11:54 +00:00
Edward Hervey
d94535832b gst-libs: Add --warn-all to introspection scanner
And let's get fixing those docs :)
2011-11-25 10:31:38 +01:00
Vincent Penquerc'h
44ec58b41b libgstriff: add a couple tags that need skipping
Found in a sample in the wild, appears to be ID3 tag.

https://bugzilla.gnome.org/show_bug.cgi?id=660249
2011-11-24 17:04:19 +00:00
Matej Knopp
817f39608c Fix printf format compiler warnings for OSX / 64bit
https://bugzilla.gnome.org/show_bug.cgi?id=662607
2011-11-22 01:00:59 +00:00
Wim Taymans
8fc2a21775 update for activation changes 2011-11-21 13:35:34 +01:00
Wim Taymans
d0bd5f04c0 update for new scheduling query 2011-11-18 17:58:58 +01:00
Wim Taymans
1ad4d20607 add parent to activate functions 2011-11-18 13:56:04 +01:00
Wim Taymans
285702a1a6 fix for scheduling mode rename 2011-11-18 12:37:10 +01:00
Wim Taymans
7afdff3575 Merge branch 'master' into 0.11
Conflicts:
	gst-libs/gst/audio/gstaudiodecoder.c
2011-11-17 17:07:41 +01:00
Wim Taymans
3a53451501 tag: update for new typefind 2011-11-17 16:15:46 +01:00
Wim Taymans
e302833e65 add parent to pad functions 2011-11-17 12:48:25 +01:00
Mark Nauwelaerts
69c2c46472 audioencoder: invalidate format info when setup negotiation failed
... which ensures nothing subsequently tries to slip past _chain
and into a possibly improperly setup subclass.
2011-11-16 19:03:47 +01:00
Vincent Penquerc'h
f17f918b75 audiodecoder: accept dropped buffers before we know the format
This allows flacdec to not emit audio for headers, while allowing
the base audio decoder to keep its timestamps in sync.
2011-11-16 16:54:03 +00:00
Wim Taymans
2202511e77 add parent to query function 2011-11-16 17:25:17 +01:00
Wim Taymans
28157e6f21 _query_peer_*() -> _peer_query_*() 2011-11-15 18:04:17 +01:00
Wim Taymans
026ec68f75 _peer_get_caps() -> _peer_query_caps() 2011-11-15 18:04:17 +01:00
Wim Taymans
7402d3a3d2 update for _get_caps() -> _query_caps() 2011-11-15 18:04:17 +01:00
Wim Taymans
ab9ffa93f5 change getcaps to query
Add sink and src event functions in rtpbasepayload
Add query vmethod to rtpbasepayload.
2011-11-15 18:04:16 +01:00
Vincent Penquerc'h
3e095382a1 audiodecoder: accept dropped buffers before we know the format
This allows flacdec to not emit audio for headers, while allowing
the base audio decoder to keep its timestamps in sync.
2011-11-15 13:29:31 +00:00
Robert Swain
a23dff1fbb audio: Remove some unused variables 2011-11-14 12:49:50 +01:00
Olivier Crête
82827df405 rtcpbuffer: Add feedback message types from RFC 5104
These are Codec Control messages (CCM)

https://bugzilla.gnome.org/show_bug.cgi?id=658419
2011-11-14 12:24:16 +01:00
Mark Nauwelaerts
38615abdd8 audiodecoder: improve reverse playback
... by doing some more (reverse) timestamp interpolating and
refactoring downstream pushing.

Fixes #661983.
2011-11-14 12:00:06 +01:00
Tim-Philipp Müller
046dc1097c tag: convert GstTagDemux's sometimes source pad to an always source pad
Originally decodebin couldn't deal with that in 0.10, but now simply
setting the caps when we know them should be enough. Pad activation
mode switching might need some more testing/tweaking with the new
arrangement.
2011-11-14 10:07:06 +00:00
Wim Taymans
fc04bcecbe fix docs 2011-11-14 10:46:56 +01:00
Tim-Philipp Müller
c76e5804b3 Update for GstURIHandler get_protocols() changes 2011-11-13 23:44:23 +00:00
Tim-Philipp Müller
455f337e3d gio, appsrc, appsink, cdaudiosrc: update for GstURIHandler API changes 2011-11-13 18:22:06 +00:00
Tim-Philipp Müller
4b0dce5148 Merge remote-tracking branch 'origin/master' into 0.11
Conflicts:
	gst-libs/gst/audio/Makefile.am
	gst-libs/gst/audio/audio.h
	tests/examples/seek/jsseek.c
	tests/examples/seek/seek.c
	tests/icles/test-colorkey.c
2011-11-13 13:36:29 +00:00
Tim-Philipp Müller
cd21e69913 audio: add GST_AUDIO_INFO_IS_VALID macro and use in audio decoder base class
API: GST_AUDIO_INFO_IS_VALID
2011-11-13 13:18:16 +00:00
Tim-Philipp Müller
394b1f8c3c audio: fix order in LIBADD
Local libs must come first.
2011-11-12 12:13:05 +00:00
Tim-Philipp Müller
756c9e2948 audio: fix order in LIBADD
Local libs must come first.
2011-11-12 11:58:59 +00:00
Tim-Philipp Müller
dfc13ec632 cdda: rename GstCddaBaseSrc to GstAudioCdSrc and move to libgstaudio
Another mini-lib down, to make space for new mini libs.

Remove bogus copyright line while at it.
2011-11-12 11:58:58 +00:00
René Stadler
8023733da8 video: init chroma-size and colorimetry members even if missing from caps
This makes a TRUE return from gst_video_info_from_caps fully consistent with
gst_video_info_init.
2011-11-11 19:57:25 +01:00
Wim Taymans
bdf3845498 rtsp: cleanup headers
Add padding, fix indentation, remove deprecated stuff
2011-11-11 19:35:33 +01:00
Wim Taymans
107d5a3d05 rtp: fix headers
indent, add padding, remove old abidata
2011-11-11 19:21:09 +01:00
Wim Taymans
370dca92d5 remove padding from interfaces 2011-11-11 19:16:54 +01:00
Wim Taymans
43eafea6f6 fix docs 2011-11-11 19:16:12 +01:00
Wim Taymans
fa897def26 mixertrack: fix docs 2011-11-11 19:14:26 +01:00
Wim Taymans
c42e257751 audio: fix docs 2011-11-11 19:13:52 +01:00
Wim Taymans
bfd417644d pbutils: clean up headers
Add padding
indent
2011-11-11 19:01:56 +01:00
Wim Taymans
7fb914d5b6 interfaces: clean up
Remove deprecated bits
Fix FIXMES
Indent
Add padding
2011-11-11 18:49:09 +01:00
Wim Taymans
40be2eec9f fft: fix headers
More fft structure into .c file
indent headers
2011-11-11 18:23:22 +01:00
Wim Taymans
b645287775 audio: fix headers
Add const to some methods.
Add padding.
Add GType for GstAudioInfo and GstAudioFormatInfo.
Add new/copy/free for GstAudioInfo.
2011-11-11 17:53:03 +01:00
Wim Taymans
b12aabc9da app: fix headers 2011-11-11 17:52:36 +01:00
Wim Taymans
06a6ab3e32 video: add support for max-framerate
Add support for max-framerate in the video helpers and update the video
caps document.
2011-11-11 13:14:21 +01:00
Wim Taymans
b14e3b9adc remove bogus file 2011-11-11 12:35:50 +01:00
Wim Taymans
5f1312b5d8 rename files to match object names 2011-11-11 12:32:23 +01:00
Wim Taymans
ccf511a5d4 rename BaseRTP -> RTPBase 2011-11-11 12:24:08 +01:00
Wim Taymans
a3416bc11f rename baseaudio* -> audiobase* 2011-11-11 12:00:52 +01:00
Wim Taymans
ee7072fe7e rename GstBaseAudio* ->GstAudioBase* 2011-11-11 11:52:47 +01:00
Wim Taymans
3d0ac3ded2 rename files to match contained objects 2011-11-11 11:33:15 +01:00
Wim Taymans
6511f36fdb audio: GstRingBuffer -> GstAudioRingBuffer 2011-11-11 11:21:41 +01:00
Wim Taymans
b81af23992 audio: rename internal audio ringbuffer 2011-11-11 10:54:39 +01:00
Wim Taymans
ad8f694ec6 remove bogus files
They got somehow commited in 7012e88090
2011-11-11 10:39:52 +01:00
Wim Taymans
e338792ab0 update for adapter api changes 2011-11-10 18:32:39 +01:00
Wim Taymans
fe766cf9f4 videosink: reset padding 2011-11-10 17:52:36 +01:00
Wim Taymans
ace51b689f rtsp: remove deprecated base64 library 2011-11-10 17:39:10 +01:00
Wim Taymans
f8ef57ca48 Merge branch 'master' into 0.11 2011-11-10 17:26:12 +01:00
Wim Taymans
24347217a5 rtp: fix de/payloaders
gst_basertppayload -> gst_base_rtp_payload
Add pts/dts support in the depayloader
Remove old timestamp code
Add a default getcaps function so subclasses can chain up to it instead of
relying on the return value of the getcaps function.
2011-11-10 17:18:00 +01:00
Vincent Penquerc'h
0d47c615ad baseaudiosink: make unsigned properties unsigned, not signed 2011-11-10 15:55:31 +00:00
Wim Taymans
57eaf388e0 audio: fix base class vmethods 2011-11-10 16:24:12 +01:00
Wim Taymans
ea9bc40bf9 audiosrc: avoid deadlock 2011-11-10 16:05:19 +01:00
Wim Taymans
1f8fe283f6 audioclock: remove _full version 2011-11-10 13:51:23 +01:00
Wim Taymans
f80d73468e appsink: fix header 2011-11-10 13:51:23 +01:00
Edward Hervey
3fa654b41c pbutils: Fix introspection annotations
Fixes #663689
2011-11-10 12:47:51 +01:00
Wim Taymans
d77c8cafee Merge branch 'master' into 0.11
Conflicts:
	common
	ext/pango/gsttextoverlay.c
	gst-libs/gst/video/video.c
2011-11-09 12:11:59 +01:00
Wim Taymans
372b9329b9 remove query types 2011-11-09 11:47:54 +01:00
Wim Taymans
308f6301a8 update for pad probe api changes 2011-11-08 11:08:21 +01:00
Stefan Sauer
e9629e37b7 video: log important details and fix format strings
If we complain about wrong parameters passed, also log the actual value.
2011-11-08 09:32:00 +01:00
Tim-Philipp Müller
d7fc45f42e docs: fix up some Since: markers 2011-11-07 23:05:44 +00:00
Wim Taymans
616e9b706e fix for new pad probe types
Restore the previous behaviour by only blocking downstream items and not
upstream events.
2011-11-07 17:10:48 +01:00
Wim Taymans
7ac25e9b26 Merge branch 'master' into 0.11
Conflicts:
	common
	configure.ac
	gst-libs/gst/audio/gstbaseaudiosink.c
	gst/playback/gstdecodebin2.c
	gst/playback/gstplaysinkaudioconvert.c
	gst/playback/gstplaysinkaudioconvert.h
	gst/playback/gstplaysinkvideoconvert.c
	gst/playback/gstplaysinkvideoconvert.h
2011-11-07 12:23:15 +01:00
Felipe Contreras
3df415d4c7 baseaudiosink: make discont-wait configurable
Now we can configure how much time to wait before deciding that a
discont has happened.

Also, adds getter and setter to allow derived implementations to set
this value upon construction.

Suggestions and several improvements by Havard Graff.

Signed-off-by: Felipe Contreras <felipe.contreras@gmail.com>
2011-11-07 11:58:46 +01:00
Felipe Contreras
0a111bf26e baseaudiosink: delay the resyncing of timestamp vs ringbuffertime
A common problem for audio-playback is that the timestamps might not
be completely linear. This is specially common when doing streaming over
a network, where you can have jittery and/or bursty packettransmission,
which again will often be reflected on the buffertimestamps.

Now, the current implementation have a threshold that says how far the
buffertimestamp is allowed o drift from the ideal aligned time in the
ringbuffer. This was an instant reaction, and ment that if one buffer
arrived with a timestamp that would breach the drift-tolerance, a resync
would take place, and the result would be an audible gap for the
listener.

The annoying thing would be that in the case of a "timestamp-outlier",
you would first resync one way, say +100ms, and then, if the next
timestamp was "back on track", you would end up resyncing the other way
(-100ms) So in fact, when you had only one buffer with slightly off
timestamping, you would end up with *two* audible gaps. This is the
problem this patch addresses.

The way to "fix" this problem with the previous implementation, would
have been to increase the "drift-tolerance" to a value that was greater
than the largest timestamp-outlier one would normally expect.  The big
problem with this approach, however, is that it will allow normal
operations with a huge offset timestamp vs running-time, which is
detrimental to lip-sync. If the drift-tolerance is set to 200ms, it
basically means that lip-sync can easily end up being off by that much.

This patch will basically start a timer when the first breach of
drift-tolerance is detected. If any following timestamp for the next n
nanoseconds gets "back on track" within the threshold, it has basically
eliminated the effect of an outlier, and the timer is stopped.  If,
however, all timestamps within this time-limit are breaching the
threshold, we are probably facing a more permanent offset in the
timestamps, and a resync is allowed to happen.

So basically this patch offers something as rare as both higher
accuracy, it terms of allowing smaller drift-tolerances, as well as much
smoother, less glitchy playback!

Commit message and improvments by Havard Graff.

Fixes bug #640859.
2011-11-07 11:33:32 +01:00
Felipe Contreras
3f1395afae baseaudiosink: rename some variables 2011-11-07 11:18:34 +01:00
Felipe Contreras
fbde258be6 baseaudiosink: use gst_util_uint64_scale_int when appropriate
It's probably safer this way.
2011-11-07 11:11:08 +01:00
Felipe Contreras
369cf3f14a baseaudiosink: split drift-tolerance into alignment-threshold
So that drift-tolerance is used for clock slaving resync, and
alignment-threshold is for timestamp drift.
2011-11-07 11:10:05 +01:00
Felipe Contreras
58b9818853 baseaudiosink: trivial comment fixes
Some found by Havard Graff.

Signed-off-by: Felipe Contreras <felipe.contreras@gmail.com>
2011-11-07 10:57:56 +01:00
Wim Taymans
2f8292b495 ringbuffer: store bpf in the right variable 2011-11-04 13:21:24 +01:00
Edward Hervey
771cbbb17c rtpbuffer: Fix compilation issues with gcc 4.6.1 2011-11-04 10:36:15 +01:00
Reynaldo H. Verdejo Pinochet
7559fb29a4 Add missing default include paths to androgenizer call
Fixes building tag/ with Android's NDK
2011-11-03 21:35:38 -03:00
Wim Taymans
f4bee46072 net: remove net library, it's now in core 2011-11-03 16:48:51 +01:00
Wim Taymans
a5fa136c0b update for tag API removal 2011-11-02 12:11:16 +01:00
Edward Hervey
dfc9d1658d video: Add convenience macros for accessing GstVideoInfo flags 2011-11-02 11:24:33 +01:00
Wim Taymans
4e6563d91c netbuffer: _netaddress_ -> _net_address_ 2011-11-02 09:04:28 +01:00
Wim Taymans
e2015eeb5f netaddress: updata api 2011-11-02 09:04:27 +01:00
Wim Taymans
e067e67923 rename meta* -> *meta 2011-11-02 09:04:27 +01:00
Wim Taymans
5bdfd6d899 structure: fix for api update 2011-11-02 09:04:27 +01:00
Wim Taymans
df4999aeb1 bufferlist: update for new API 2011-11-02 09:04:27 +01:00
Tim-Philipp Müller
b52c5819fb Update for pad API changes
GstProbeType, GstProbeReturn and GstActivateMode -> GstPad*
2011-11-01 00:34:28 +00:00
Tim-Philipp Müller
220ccdf275 audioencoder: save audio info parsed in setcaps in encoder context
Otherwise we'll just error out when the first buffer gets pushed.
This is a porting artefact, in 0.10 the infos were allocated on the
heap, now we're doing everything with stack-allocated structs.
2011-10-31 14:22:39 +00:00
Tim-Philipp Müller
5ee51e47a1 ext, gst, gst-libs, tests: update for tag list API changes 2011-10-31 14:22:39 +00:00
René Stadler
7eb0985282 audio: remove old C file generated from template
Not sure how this one got pulled into a merge. In 0.10, it was moved away to
gst-template a long time ago. gstaudiofilterexample.c got generated from
gstaudiofiltertemplate.c.
2011-10-31 15:19:54 +01:00
Wim Taymans
95281cc306 Merge branch 'master' into 0.11 2011-10-28 16:24:44 +02:00
Wim Taymans
7247eb5f2c fix compile for SEEK_TYPE_CUR removal 2011-10-28 16:11:36 +02:00
Mersad Jelacic
d430eb65c5 audiosink: avoid deadlocking audioringbuffer thread
... when it goes into wait for ringbuffer starting just after such
having been signalled.

Fixes #661738.
2011-10-28 14:07:40 +02:00
Wim Taymans
b70275fa10 audiofilter: use BPF for unit_size 2011-10-28 11:37:31 +02:00
René Stadler
9beff28579 audiofilter: fix get_unit_size 2011-10-28 11:24:00 +02:00
René Stadler
5d2154ff4b audiofilter: init audio info sooner 2011-10-28 11:24:00 +02:00
René Stadler
372cf41a6d audio, video: init audio/video format info to UNKNOWN format
This is to prevent e.g. GST_AUDIO_INFO_FORMAT() from crashing on a NULL pointer
dereference when used with an unset info.
2011-10-28 11:24:00 +02:00
Wim Taymans
01854cca80 basertppay: rename caps fields
Make the caps fields for timestamp and seqnum match the element
properties.

See #628773
2011-10-27 18:54:50 +02:00
Wim Taymans
9555229e79 basedepay: remove old fields 2011-10-27 18:50:32 +02:00
Wim Taymans
06311362e9 fix compilation 2011-10-27 17:26:58 +02:00
Wim Taymans
016d036137 Merge branch 'master' into 0.11
Conflicts:
	configure.ac
	gst-libs/gst/audio/gstbaseaudiosink.c
	gst/audioconvert/channelmixtest.c
	gst/playback/gstplaybasebin.c
	gst/playback/gstsubtitleoverlay.c
	tests/examples/Makefile.am
	tests/examples/audio/Makefile.am
2011-10-27 15:44:58 +02:00
Wim Taymans
b21bb37657 overlay: fix compilation 2011-10-27 15:29:36 +02:00
Stefan Sauer
53d7d2e966 interfaces: clean up the use of iface and class/klass 2011-10-21 14:46:48 +02:00
Mark Nauwelaerts
981070eb44 audiodecoder: having gather queue contents implies some draining is in order
... which ensures e.g. processing and sending last fragment of reverse playback
downstream at EOS.
2011-10-19 16:51:09 +02:00
Tim-Philipp Müller
4e59e63ff7 baseaudiosink: fix unused variable compiler warning if debugging in core is disabled
https://bugzilla.gnome.org/show_bug.cgi?id=660150
2011-10-19 00:32:13 +01:00
Edward Hervey
12a8fff8ac audio: Add some default channel positions 2011-10-17 12:00:55 +02:00
Edward Hervey
b4858253dc audio: Properly handle signedness in gst_audio_format_build_integer() 2011-10-17 12:00:16 +02:00
Edward Hervey
45c4a19472 audio: Indent and doc fixes 2011-10-17 11:45:39 +02:00
Edward Hervey
8268a7a20e discoverer: Only call gst_video_info_from_caps on raw video 2011-10-11 17:42:35 +02:00
Wim Taymans
f1088ed647 update for UNEXPECTED -> EOS flowreturn 2011-10-10 11:39:52 +02:00
Thiago Santos
123671bc05 libs: video: Add protection against null strings
Check and assert if input for gst_video_format_from_string is null.
Return GST_VIDEO_FORMAT_UNKNOWN as a fallback
2011-10-09 17:05:15 -03:00
Tim-Philipp Müller
ab949eebbd audiodecoder: update to 0.11 API after merge 2011-10-09 16:15:54 +01:00
Tim-Philipp Müller
303dbaf84b Merge remote-tracking branch 'origin/master' into 0.11
Conflicts:
	tests/check/pipelines/vorbisdec.c
	tests/check/pipelines/vorbisenc.c
2011-10-09 16:08:36 +01:00
Alessandro Decina
bc6f00becb audioencoder: fix compile warning 2011-10-09 16:48:18 +02:00
Mark Nauwelaerts
871b1584c9 audioencoder: only resync to upstream upon discont in perfect ts mode
... as documented, where discont is marked here if tolerance has been
exceeded.
2011-10-08 20:20:10 +02:00
Mark Nauwelaerts
a7ce550d04 audiodecoder: fix timestamp tolerance handling 2011-10-08 20:20:06 +02:00
Mark Nauwelaerts
d8312994aa audiodecoder: handle empty input by discarding 2011-10-08 20:20:03 +02:00
Wim Taymans
73b894107a Merge branch 'master' into 0.11
Conflicts:
	ext/vorbis/gstvorbisdec.c
	ext/vorbis/gstvorbisenc.c
	ext/vorbis/gstvorbisenc.h
	gst/audiotestsrc/gstaudiotestsrc.c
2011-10-08 10:19:06 +02:00
Mark Nauwelaerts
37c629fcc6 audioencoder: make upstream queries MT-safe 2011-10-07 14:52:50 +02:00
Mark Nauwelaerts
77069f01b1 audiodecoder: make upstream queries and events MT-safe 2011-10-07 14:52:48 +02:00
Edward Hervey
b8219faa90 audio: Make sure 'channels' and 'channel-positions' are coherent
If channel-positions are present, check they match the reported
'channels' value.
2011-10-05 11:57:54 +02:00
Edward Hervey
70d967da7c audio: Fix overread in channel positions
The array we're writing to is limited to 64 ... but the amount of
input positions might be lower than 64. Therefore use MIN and not
MAX to know how many values to read from the array.
2011-10-05 11:51:07 +02:00
Wim Taymans
a00927ad03 Merge branch 'master' into 0.11 2011-10-04 17:58:49 +02:00
Vincent Penquerc'h
93900d47ed encoding-profile: add a function to create a profile from a discoverer info
Only A/V streams are added at the moment, there does not seem to be
a similar way to add other streams (eg, subtitles).

https://bugzilla.gnome.org/show_bug.cgi?id=642878
2011-10-03 11:51:23 +02:00