Wim Taymans
47cbb230e9
audio: move audio interfaces
...
Move the audio related interfaces to the audio library.
2011-11-30 07:57:02 +01:00
Tim-Philipp Müller
0c056a04fe
Merge commit '4a58223e4c824fedc024af435337a769e8ce593e' into 0.11
2011-11-28 21:20:10 +00:00
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
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
Matej Knopp
817f39608c
Fix printf format compiler warnings for OSX / 64bit
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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
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
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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
ab9ffa93f5
change getcaps to query
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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
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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
Mark Nauwelaerts
38615abdd8
audiodecoder: improve reverse playback
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... by doing some more (reverse) timestamp interpolating and
refactoring downstream pushing.
Fixes #661983 .
2011-11-14 12:00:06 +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
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Local libs must come first.
2011-11-12 12:13:05 +00:00
Tim-Philipp Müller
756c9e2948
audio: fix order in LIBADD
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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
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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
Wim Taymans
c42e257751
audio: fix docs
2011-11-11 19:13:52 +01:00
Wim Taymans
b645287775
audio: fix headers
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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
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
f8ef57ca48
Merge branch 'master' into 0.11
2011-11-10 17:26:12 +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
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
Tim-Philipp Müller
d7fc45f42e
docs: fix up some Since: markers
2011-11-07 23:05:44 +00:00
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
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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
Wim Taymans
a5fa136c0b
update for tag API removal
2011-11-02 12:11:16 +01:00
Wim Taymans
5bdfd6d899
structure: fix for api update
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
Mersad Jelacic
d430eb65c5
audiosink: avoid deadlocking audioringbuffer thread
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... 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
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
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
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... 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
Wim Taymans
f1088ed647
update for UNEXPECTED -> EOS flowreturn
2011-10-10 11:39:52 +02:00
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
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... 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
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
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