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
Wim Taymans
8778cff7f0
video: add h264 transfer functions
2011-10-03 10:02:43 +02:00
Tim-Philipp Müller
6ec5fc8d95
audio: don't use GST_PTR_FORMAT for segments
...
Avoids crashes with debugging output enabled.
2011-09-30 10:56:02 +01:00
Wim Taymans
67f1a097bf
video: add another color matrix for mpeg2
2011-09-30 11:04:19 +02:00
Wim Taymans
9592796d8a
video: fix docs
2011-09-30 11:04:19 +02:00
Wim Taymans
1395378575
audiodecoder: fix refcounting error
2011-09-28 16:08:14 +02:00
Wim Taymans
ca6ebee870
ringbuffer: store info so we can debug it
2011-09-28 16:07:53 +02:00
Wim Taymans
f97a9bdc68
Merge branch 'master' into 0.11
2011-09-28 15:46:40 +02:00
Mark Nauwelaerts
8633eb391d
audiodecoder: really push pending events
2011-09-28 15:42:46 +02:00
Wim Taymans
19626cf27a
audiodecoder: add method to set output caps
...
Add a method to configure the output caps. Subclasses can't use
gst_pad_set_caps() anymore because then we won't see the caps.
Unbreak the padtemplate registration, the GTypeClass that is configured in the
object during _init is not the right one, we need to use the klass passed as the
argument to the init function..
2011-09-28 15:35:56 +02:00
Tim-Philipp Müller
e4e2e3c7b0
audioencoder: remove more tags from upstream tag events such as bitrate tags
...
We want to remove all codec specific tags.
2011-09-28 14:32:20 +01:00
Wim Taymans
19346c2c3b
Merge branch 'master' into 0.11
...
Conflicts:
gst-libs/gst/audio/gstaudioencoder.c
gst/playback/gstplaybin2.c
gst/videotestsrc/videotestsrc.c
2011-09-28 11:35:46 +02:00
Mark Nauwelaerts
01d27ee084
audioencoder: only got_data if we really got some
...
... which avoids going loopy with casual subclass.
2011-09-27 16:58:44 +02:00
Mark Nauwelaerts
24d71cf7a6
audioencoder: really push pending events
2011-09-27 16:58:41 +02:00
Mark Nauwelaerts
803b65613b
audioencoder: send tag event after pending events
...
... which probably includes a pending newsegment event.
2011-09-27 16:21:55 +02:00
Mark Nauwelaerts
89f6720545
audioencoder: protect pending_events with proper lock
2011-09-27 16:21:45 +02:00
Mark Nauwelaerts
9a9541ff35
audioencoder: clean up some documentation
2011-09-27 16:21:41 +02:00
Wim Taymans
4bf9022e0c
docs: improve docs
2011-09-27 11:19:24 +02:00
Wim Taymans
c290b8044a
audioenc: fix compilation
2011-09-26 21:11:14 +02:00
Wim Taymans
f71511edd2
Merge branch 'master' into 0.11
...
Conflicts:
gst-libs/gst/audio/gstaudiodecoder.c
gst-libs/gst/audio/gstaudioencoder.c
gst/encoding/gstencodebin.c
2011-09-26 19:22:05 +02:00
Sebastian Dröge
e4c895dfaf
audioencoder: Improve set_frame_sample_{min,max} documentation
2011-09-26 16:35:55 +02:00
Sebastian Dröge
b767be2f68
audiodecoder: Fix thread safety issues if both pads have different streaming threads
2011-09-26 16:22:00 +02:00
Sebastian Dröge
d0bf465248
audiodecoder: Delay sending of serialized events to finish_frame()
2011-09-26 16:19:42 +02:00
Sebastian Dröge
f3f416004f
Revert "audioencoder: Use GST_BOILERPLATE instead of custom GObject boilerplate code"
...
This reverts commit 11e375486e
.
GST_BOILERPLATE() can't define an abstract type and
G_DEFINE_ABSTRACT_TYPE() does not pass the class struct to
the instance_init function and there's no way to get the
class struct of the current type in instance_init().
2011-09-26 16:02:51 +02:00
Sebastian Dröge
4fa9749106
audioencoder: Add support for requesting a minimum and maximum number of samples per frame
...
This extends the special case of a fixed number of samples per frame
that was supported before already.
2011-09-26 15:59:22 +02:00
Sebastian Dröge
16c3d6b3d5
audioencoder: Fix thread safety issues if both pads have different streaming threads
2011-09-26 15:45:40 +02:00
Sebastian Dröge
61ffd7cb42
audioencoder: Delay sending of serialized events to finish_frame()
...
This makes sure that the caps are already set before any serialized
events are sent downstream.
2011-09-26 15:42:14 +02:00
Sebastian Dröge
11e375486e
audioencoder: Use GST_BOILERPLATE instead of custom GObject boilerplate code
2011-09-26 15:34:54 +02:00
Mark Nauwelaerts
abafb030ac
audioencoder: add some tag handling convenience help
2011-09-26 15:15:03 +02:00
Mark Nauwelaerts
a99b313c26
audioencoder: provide CODEC/AUDIO_CODEC handling
2011-09-26 15:10:08 +02:00
Mark Nauwelaerts
aae0312e10
audioencoder: filter AUDIO_CODEC/CODEC tags from passing tag events
2011-09-26 15:10:06 +02:00
Tim-Philipp Müller
754b22d7ee
libs: remove unused floatcast header-only library
...
There's no code whatsoever that uses these macros. If anyone
ever feels the need to resurrect them, we should add them to
gstutils.h in core or libgstaudio or so.
2011-09-23 21:18:47 +01:00
Edward Hervey
17bfba09f1
Merge branch 'master' into 0.11
...
Conflicts:
ext/ogg/gstoggdemux.c
ext/pango/gsttextoverlay.c
gst-libs/gst/audio/gstaudioencoder.c
gst-libs/gst/audio/gstbaseaudiosrc.c
gst/playback/gstsubtitleoverlay.c
gst/videorate/gstvideorate.c
2011-09-23 18:27:11 +02:00
Edward Hervey
3f45eb1cfc
gst-libs: Temporarily remove dependency of gstaudio on gstpbutils
...
Also re-order the SUBDIRS in the higher-level Makefile so it cleanly
installs.
https://bugzilla.gnome.org/show_bug.cgi?id=657675
2011-09-23 16:17:45 +02:00
Mark Nauwelaerts
001b4a0072
audioencoder: proxy some more optional downstream caps fields to upstream
2011-09-22 15:47:06 +02:00
Mark Nauwelaerts
2a362a95f7
audioencoder: changed is verily the opposite of equal
2011-09-22 15:47:06 +02:00
Mark Nauwelaerts
b420dd54ea
audioencoder: prevent crashing when comparing to a freshly inited GstAudioInfo
2011-09-22 15:46:56 +02:00
Mark Nauwelaerts
7fa7de9221
audio: some more accessor macros for GstAudioInfo
2011-09-22 15:45:05 +02:00
Mark Nauwelaerts
b44978befe
audiodecoder: fix documentation typo
2011-09-22 15:45:01 +02:00
Age Bosma
043ee22e25
discoverer: Don't use gtk-doc /* < ... > */ style comments for signals
...
The /*< ... >*/ style is only used for public|protected|private,
signal comments use /* signals */. This prevents the some code
parsers/binding generators to be confused by the comment.
2011-09-19 14:36:00 +02:00
Mark Nauwelaerts
e574f58e71
rtspdefs: add RTCP-Interval header
2011-09-19 11:32:23 +02:00
Tim-Philipp Müller
454c554b11
docs: minor addition to GST_TAG_ID3V2_HEADER_SIZE docs
2011-09-12 19:55:40 +01:00
Tim-Philipp Müller
55182ed841
baseaudiosrc: don't try to fixate "width" field for alaw/mulaw
...
Fixes warning when trying to fixate e.g. pulsesrc ! audio/x-alaw ! fakesink.
2011-09-10 18:30:55 +01:00
Tim-Philipp Müller
0f38f86182
colorbalance: add some guards to interface methods
...
https://bugzilla.gnome.org/show_bug.cgi?id=658584
2011-09-09 13:09:43 +01:00
Tim-Philipp Müller
4529c6dc32
Merge remote-tracking branch 'origin/master' into 0.11
...
Merge in doc updates for audio enums from 0.10, and get rid
of the #if #else in the enum list, since that confuses gtk-doc.
Conflicts:
gst-libs/gst/audio/audio.c
gst-libs/gst/audio/audio.h
2011-09-06 16:42:42 +01:00
Wim Taymans
dc28bd1b63
audio: rename IS_LE/BE to _IS_LITTLE_ENDIAN/BIG_ENDIAN
2011-09-06 16:27:27 +01:00
Wim Taymans
f04b8fd8af
audio/video add descriptions
...
Add a description to the audio and video format info in case we want to use this
later.
2011-09-06 16:46:48 +02:00
Tim-Philipp Müller
36a75bdb71
audio: update internal silent sample defines as well to match 0.11
2011-09-06 15:46:45 +01:00
Wim Taymans
c0d31dd555
rename IS_LE/BE to _IS_LITTLE_ENDIAN/BIG_ENDIAN
2011-09-06 16:46:02 +02:00
Tim-Philipp Müller
91d1112360
audio: update audio format enums to match changes in 0.11
...
And add new audio format info stuff to docs.
2011-09-06 15:36:51 +01:00
Wim Taymans
7012e88090
Merge branch 'master' into 0.11
...
Conflicts:
gst-libs/gst/audio/audio.h
gst-libs/gst/audio/gstaudiodecoder.c
gst-libs/gst/audio/gstaudiodecoder.h
gst-libs/gst/audio/gstaudioencoder.c
gst-libs/gst/audio/gstbaseaudioencoder.h
gst/playback/Makefile.am
gst/playback/gstplaybin.c
gst/playback/gstplaysink.c
gst/playback/gstplaysinkvideoconvert.c
gst/playback/gstsubtitleoverlay.c
gst/videorate/gstvideorate.c
gst/videoscale/gstvideoscale.c
win32/common/libgstaudio.def
2011-09-06 15:24:32 +02:00
Wim Taymans
33196cdd2c
audio: change audio format syntax a little
...
Remove the _ in front of the endianness prefix.
Remove the _3 postfix for the 24 bits formats.
Add a _32 postfix after the formats that occupy extra space beyond their
natural size.
The result is that the GST_AUDIO_NE() macro can simply append the endianness
after all formats and that we only specify a different sample width when it is
different from the natural size of the sample. This makes things more consistent
and follows the pulseaudio conventions instead of the alsa ones.
2011-09-06 12:06:39 +02:00
Tim-Philipp Müller
9a8a989a22
docs: more docs clean-ups
2011-09-06 10:07:33 +01:00
Tim-Philipp Müller
5e61db25b5
audio: fix GST_AUDIO_FORMAT_INFO_IS_*() macros to return a boolean
2011-09-05 23:28:20 +01:00
Tim-Philipp Müller
ba05716485
docs: some docs love
2011-09-05 23:28:20 +01:00
Tim-Philipp Müller
7563e0c9cf
docs: add GstAudioDecoder and GstAudioEncoder to documentation
2011-09-05 23:28:20 +01:00
Tim-Philipp Müller
86e6343759
audio: rename GstBaseAudioDecoder/Encoder to GstAudioDecoder/Encoder
...
API: gst_gst_audio_decoder_finish_frame()
API: gst_gst_audio_decoder_get_audio_info()
API: gst_gst_audio_decoder_get_byte_time()
API: gst_gst_audio_decoder_get_delay()
API: gst_gst_audio_decoder_get_latency()
API: gst_gst_audio_decoder_get_max_errors()
API: gst_gst_audio_decoder_get_min_latenc()y
API: gst_gst_audio_decoder_get_parse_state()
API: gst_gst_audio_decoder_get_plc()
API: gst_gst_audio_decoder_get_plc_aware()
API: gst_gst_audio_decoder_get_tolerance()
API: gst_gst_audio_decoder_get_type()
API: gst_gst_audio_decoder_set_byte_time()
API: gst_gst_audio_decoder_set_latency()
API: gst_gst_audio_decoder_set_max_errors()
API: gst_gst_audio_decoder_set_min_latency()
API: gst_gst_audio_decoder_set_plc()
API: gst_gst_audio_decoder_set_plc_aware()
API: gst_gst_audio_decoder_set_tolerance()
API: gst_gst_audio_encoder_finish_frame()
API: gst_gst_audio_encoder_get_audio_info()
API: gst_gst_audio_encoder_get_frame_max()
API: gst_gst_audio_encoder_get_frame_samples()
API: gst_gst_audio_encoder_get_hard_resync()
API: gst_gst_audio_encoder_get_latency()
API: gst_gst_audio_encoder_get_lookahead()
API: gst_gst_audio_encoder_get_mark_granule()
API: gst_gst_audio_encoder_get_perfect_timestamp()
API: gst_gst_audio_encoder_get_tolerance()
API: gst_gst_audio_encoder_get_type()
API: gst_gst_audio_encoder_proxy_getcaps()
API: gst_gst_audio_encoder_set_frame_max()
API: gst_gst_audio_encoder_set_frame_samples()
API: gst_gst_audio_encoder_set_hard_resync()
API: gst_gst_audio_encoder_set_latency()
API: gst_gst_audio_encoder_set_lookahead()
API: gst_gst_audio_encoder_set_mark_granule()
API: gst_gst_audio_encoder_set_perfect_timestamp()
API: gst_gst_audio_encoder_set_tolerance()
https://bugzilla.gnome.org/show_bug.cgi?id=642690
2011-09-05 23:28:13 +01:00
Wim Taymans
2f2aa4ac32
video: improve docs a little
2011-08-31 13:32:21 +02:00
Wim Taymans
de4aeab544
video: add some more macros
2011-08-30 14:04:54 +02:00
Wim Taymans
e694528155
base: port to 0.11
2011-08-29 13:28:08 +02:00
Wim Taymans
057aecc34e
audio: fix after merge
2011-08-29 11:42:35 +02:00
Wim Taymans
dc2b00adb8
pbutils: port to new API
2011-08-29 11:38:01 +02:00
Wim Taymans
e1287b97ab
Merge branch 'master' into 0.11
...
Conflicts:
ext/ogg/gstoggmux.c
gst-libs/gst/audio/audio.c
gst-libs/gst/audio/audio.h
gst-libs/gst/audio/multichannel.h
gst-libs/gst/pbutils/Makefile.am
gst-libs/gst/pbutils/gstdiscoverer.c
gst/playback/gstplaysinkaudioconvert.c
gst/playback/gstplaysinkvideoconvert.c
win32/common/libgstaudio.def
2011-08-29 11:37:36 +02:00
Tim-Philipp Müller
67a12c9c72
pbutils: don't depend on libgstvideo just to parse some caps
...
Let's extract those ints and fractions ourselves and not depend
on libgstvideo.
2011-08-27 14:57:41 +01:00
Tim-Philipp Müller
517153e85a
audio: add GstBaseAudioDecoder and GstBaseAudioEncoder to build
...
However, libgstaudio now depends on libgstvideo (via pbutils).
https://bugzilla.gnome.org/show_bug.cgi?id=642690
API: gst_audio_info_clear()
API: gst_audio_info_convert()
API: gst_audio_info_copy()
API: gst_audio_info_free()
API: gst_audio_info_from_caps()
API: gst_audio_info_init()
API: gst_audio_info_to_caps()
API: gst_base_audio_decoder_finish_frame()
API: gst_base_audio_decoder_get_audio_info()
API: gst_base_audio_decoder_get_byte_time()
API: gst_base_audio_decoder_get_delay()
API: gst_base_audio_decoder_get_latency()
API: gst_base_audio_decoder_get_max_errors()
API: gst_base_audio_decoder_get_min_latency()
API: gst_base_audio_decoder_get_parse_state()
API: gst_base_audio_decoder_get_plc()
API: gst_base_audio_decoder_get_plc_aware()
API: gst_base_audio_decoder_get_tolerance()
API: gst_base_audio_decoder_get_type()
API: gst_base_audio_decoder_set_byte_time()
API: gst_base_audio_decoder_set_latency()
API: gst_base_audio_decoder_set_max_errors()
API: gst_base_audio_decoder_set_min_latency()
API: gst_base_audio_decoder_set_plc()
API: gst_base_audio_decoder_set_plc_aware()
API: gst_base_audio_decoder_set_tolerance()
API: gst_base_audio_encoder_finish_frame()
API: gst_base_audio_encoder_get_audio_info()
API: gst_base_audio_encoder_get_frame_max()
API: gst_base_audio_encoder_get_frame_samples()
API: gst_base_audio_encoder_get_hard_resync()
API: gst_base_audio_encoder_get_latency()
API: gst_base_audio_encoder_get_lookahead()
API: gst_base_audio_encoder_get_mark_granule()
API: gst_base_audio_encoder_get_perfect_timestamp()
API: gst_base_audio_encoder_get_tolerance()
API: gst_base_audio_encoder_get_type()
API: gst_base_audio_encoder_proxy_getcaps()
API: gst_base_audio_encoder_set_frame_max()
API: gst_base_audio_encoder_set_frame_samples()
API: gst_base_audio_encoder_set_hard_resync()
API: gst_base_audio_encoder_set_latency()
API: gst_base_audio_encoder_set_lookahead()
API: gst_base_audio_encoder_set_mark_granule()
API: gst_base_audio_encoder_set_perfect_timestamp()
API: gst_base_audio_encoder_set_tolerance()
2011-08-27 14:47:50 +01:00
Tim-Philipp Müller
58f515f06a
docs: add since markers to baseaudio{decoder,encoder} documentation
2011-08-27 14:47:50 +01:00
Tim-Philipp Müller
90e3d25891
baseaudiodecoder, baseaudioencoder: fix some compiler warnings
...
Leaving the GST_USE_UNSTABLE_API guards in until some of the
ported decoders have been updated and it's clear that I didn't
mess up anywhere porting things to the new audio API.
2011-08-27 14:47:49 +01:00
Tim-Philipp Müller
52ecb383d7
baseaudioutils: remove, merged into or superseded by audio.c
2011-08-27 14:47:49 +01:00
Tim-Philipp Müller
7f0c7e5f82
baseaudioencoder: port to new GstAudioInfo API
2011-08-27 14:47:49 +01:00
Tim-Philipp Müller
c89b49bfaf
baseaudiodecoder: port to GstAudioInfo API
2011-08-27 14:47:49 +01:00
Tim-Philipp Müller
946ddb6462
audio: add gst_audio_info_{init,clear} and gst_audio_info_{copy,free}
2011-08-27 14:47:49 +01:00
Tim-Philipp Müller
63a3d360dc
audio: add GstAudioFormat, GstAudioFormatInfo and GstAudioInfo
...
Same as in 0.11, but with caps parsing/serialising for 0.10 style
caps. Add setting default channel positions.
2011-08-27 14:47:01 +01:00
Mark Nauwelaerts
bf4a28f420
baseaudioencoder: remove leftover experimental code
2011-08-27 14:47:00 +01:00