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Author SHA1 Message Date
Sebastian Dröge
90eb93c2ef Don't compare booleans for equality to TRUE and FALSE 
TRUE is 1, but every other non-zero value is also considered true. Comparing
for equality with TRUE would only consider 1 but not the others.
 2014-12-01 09:51:12 +01:00
Peter G. Baum
c734fbc139 audio-channels: allow partially valid channel_mask 
Since WAVEFORMATEXTENSIBLE allows to have more channels than
bits in the channel mask we should allow this, too, to avoid
loss of information.

https://bugzilla.gnome.org/show_bug.cgi?id=733405
 2014-10-14 10:29:56 +02:00
Thiago Santos
a0b25a570a audiodecoder: should post DECODE errors and not ENCODE 
Fix error code for audio decoder
 2014-10-13 22:26:29 -03:00
Arun Raghavan
c47b005197 audio: Fix up a comment in GstAudioBaseSink 
Rewrote the comment to not be PulseAudio-specific.
 2014-09-29 19:46:32 +05:30
Arun Raghavan
324ebd19e3 audio: Trivial comment for unhandled MPEG-2 payloading case 
The spec mentions a version of the MPEG-2 frame with a base frame and
extension frame. I don't have IEC 13818-3 to figure out what that is,
and don't see any references in search results, so it's a FIXME for now.

https://bugzilla.gnome.org/show_bug.cgi?id=736797
 2014-09-27 10:40:27 +05:30
Arun Raghavan
2965b796bc audio: Fixes for MPEG-2 LSF IEC61937 payloading 
The low sample frequency case for MPEG-2 is <=12kHz (the 32kHz number
applies to MPEG-1).

https://bugzilla.gnome.org/show_bug.cgi?id=736797
 2014-09-27 10:40:27 +05:30
Anuj Jaiswal
798ff6e561 audio: correct condition for MPEG case. 
Signed-off-by: Anuj Jaiswal <anuj.jaiswal@samsung.com>

https://bugzilla.gnome.org/show_bug.cgi?id=736797
 2014-09-27 10:40:27 +05:30
Thiago Santos
8242676dc2 audiosink: compensate for segment restart with clock's time_offset 
When playing chained data the audio ringbuffer is released and
then acquired again. This makes it reset the segbase/segdone
variables, but the next sample will be scheduled to play in
the next position (right after the sample from the previous media)
and, as the segdone is at 0, the audiosink will wait the duration
of this previous media before it can write and play the new data.

What happens is this:
pointer at 0, write to 698-1564, diff 698, segtotal 20, segsize 1764, base 0

it will have to wait the length of 698 samples before being able to write.

In a regular sample playback it looks like:
pointer at 677, write to 696-1052, diff 19, segtotal 20, segsize 1764, base 0

In this case it will write to the next available position and it
doesn't need to wait or fill with silence.

This solution is borrowed from pulsesink that resets the clock to
start again from 0, which makes it reset the time_offset to the time
of the last played sample. This is used to correct the place of
writing in the ringbuffer to the new start (0 again)

https://bugzilla.gnome.org/show_bug.cgi?id=737055
 2014-09-24 10:22:54 -03:00
Stefan Sauer
5f0aad6f42 audioencoder: reshuffle code in error handling 
Move the assert to the error handling block at the end of the function so the
the logging is still triggered. Reword the logging slightly and add another
comment to hint what went wrong.

Fixes #737138
 2014-09-23 11:56:33 +02:00
Sebastian Dröge
3592bd577c audiodecoder: Simplify code a bit 2014-09-18 12:40:26 +03:00
Ognyan Tonchev
2fff66b071 audioencoder: do not leak events when flushing them 
https://bugzilla.gnome.org/show_bug.cgi?id=736796
 2014-09-18 12:40:19 +03:00
Ognyan Tonchev
c674a0aa64 audiodecoder: Don't leak events 
https://bugzilla.gnome.org/show_bug.cgi?id=736788
 2014-09-17 14:11:34 +03:00
Ognyan Tonchev
add8f02703 audiocdsrc: do not leak uid after parsing TOC select event 
https://bugzilla.gnome.org/show_bug.cgi?id=736739
 2014-09-17 09:50:17 +03:00
Garg
47e303269d audiobasesink: Fix deadlock caused by holding object lock while calling clock functions 
Issue:
During a PAUSED->PLAYING transition when we are rendering an audio buffer in AudioBaseSink
we make adjustments to the sink's provided clock i.e. fix clock calibration using the external
pipeline clock, within "gst_audio_base_sink_sync_latency function inside gstaudiobasesink.c".
For the calibration adjustment we need to get the sink clock time using "gst_audio_clock_get_time".
But before calling "gst_audio_clock_get_time" we acquire the Object Lock on the Sink. If sink is
a pulsesink, "gst_audio_clock_get_time" internally calls "gst_pulsesink_get_time" which needs to
acquire Pulse Audio Main Loop Lock before querying Pulse Audio for its stream time using
"pa_stream_get_time". Please see "gst_pulsesink_get_time in pulsesink.c".

So the situation here is we have acquired the Object lock on Sink and need PA Main Loop Lock.
Now Pulse Audio Main Thread itself might be in the process of posting a stream status
message after Paused to Playing transition which in turn acquires the PA Main loop lock and
needs the Object Lock on Pulse Sink. This causes a deadlock with the earlier render thread.

Fix:
Do not acquire the object Lock on Sink before querying the time on PulseSink clock. This is
similar to the way we have used get_time at other places in the code. Acquire it after the
get_time call. This way PA Main loop will be able to post its stream status message by
acquiring the Sink Object lock and will eventually release its Main Loop lock needed for
gst_pulsesink_get_time to continue.

https://bugzilla.gnome.org/show_bug.cgi?id=736071
 2014-09-12 14:21:19 +03:00
Sebastian Dröge
d357f28260 audiodecoder: Fix broken boolean expression 
We can seek with end_type==NONE and end_type==SET && end_position=-1. The
check for end_type!=NONE made the second condition impossible.

CID 1226439
 2014-08-28 17:00:26 +03:00
Sebastian Dröge
4a69d6ba3b audiodecoder: Don't ignore ::start/stop return values 2014-08-25 13:15:07 +03:00
Jan Schmidt
02d1ab0d1c audiodecoder: Don't drain and flush on SEGMENT events. 
As was done for the base video decoder in commit 695675, don't
flush out the decoder on a new SEGMENT event. Segment events
may be a new segment, but are also often segment updates for
the current segment where the old data should be kept. For new
segments, a STREAM_START event will already trigger a drain, but
make sure to flush any remaining partial data then as well.

https://bugzilla.gnome.org/show_bug.cgi?id=734666
 2014-08-12 23:54:41 +10:00
Sebastian Rasmussen
a285f7126b audioencoder: Mark caps argument as not being transferred 
https://bugzilla.gnome.org/show_bug.cgi?id=734540
 2014-08-10 10:45:14 +01:00
Sebastian Dröge
368d75fe75 audiodecoder: Handle CAPS events immediately instead of delaying them 
https://bugzilla.gnome.org/show_bug.cgi?id=733147
 2014-07-21 09:36:00 +02:00
Sebastian Dröge
1e64667fe0 libs: There is no G_TYPE_CHECK_INTERFACE_TYPE and G_TYPE_CHECK_INTERFACE_CAST 
Remove the macros that used them, nobody could've used them anyway.
 2014-06-26 16:18:46 +02:00
Sebastian Dröge
909dd7831b audiodecoder: Don't be too picky about the output frame counter 
With most decoder libraries, and especially when accessing codecs via
OpenMAX or similar APIs, we don't have the ability to properly related
the output buffers to a number of input samples. And could e.g. get
a fractional number of input buffers decoded at a time.

Previously this would in the end lead to an error message and stopped
playback. Change it to a warning message instead and try to handle it
gracefully. In theory the subclass can now get timestamp tracking
wrong if it completely misuses the API, but if on average it behaves
correct (and gst-omx and others do) it will continue to work properly.

Also add a test for the new behaviour.

We don't change it in the encoder yet as that requires more internal logic
changes AFAIU and I'm not aware of a case where this was a problem so far.
 2014-06-20 11:02:55 +02:00
Thibault Saunier
12df7fa49d audiodecoder: Keep still meaningfull pending events on FLUSH_STOP 
Only EOS and segment should be deleted in that case.

https://bugzilla.gnome.org/show_bug.cgi?id=709868
 2014-06-03 13:03:21 +02:00
Thibault Saunier
967d1fb982 audioencoder: Keep still meaningfull pending events on FLUSH_STOP 
Only EOS and segment should be deleted in that case.

https://bugzilla.gnome.org/show_bug.cgi?id=709868
 2014-06-03 13:03:16 +02:00
Philip Withnall
ba87655628 audio: Add a missing precondition to gst_audio_format_from_string() 
https://bugzilla.gnome.org/show_bug.cgi?id=730874
 2014-05-28 11:34:01 +02:00
Thiago Santos
09b8f902ea audiodecoder: return EOS when segment is over 
if a buffer is clipped by being completely out of segment, check if this
buffer is after the end of the segment and return EOS upstream

https://bugzilla.gnome.org/show_bug.cgi?id=709224
 2014-05-26 19:26:45 -03:00
Sebastian Dröge
68f5350c66 Release 1.3.1 2014-05-03 17:50:10 +02:00
Haakon Sporsheim
7c97a1c6cf audiodecoder: Make caps writable before fixating 
https://bugzilla.gnome.org/show_bug.cgi?id=729114
 2014-04-29 09:58:21 +02:00
Tim-Philipp Müller
bcb8068e27 docs: remove outdated and pointless 'Last reviewed' lines from docs 
They are very confusing for people, and more often than not
also just not very accurate. Seeing 'last reviewed: 2005' in
your docs is not very confidence-inspiring. Let's just remove
those comments.
 2014-04-26 23:28:57 +01:00
Edward Hervey
74eb5fa995 audiodecoder: Plug caps leaks 
We were returning in various places without unreffing the caps, and
we were also leaking (overwriting) the caps we got from _get_current_caps()

Spotted by Haakon Sporsheim in #gstreamer
 2014-04-25 11:30:37 +02:00
Vincent Penquerc'h
dda777803c audiocdsrc: guard aginst overflow 
An audio CD may contain about a tenth of the samples 32 bit can
represent, so it doesn't seem likely this will be hit in practice.

Coverity 1139805
 2014-04-10 12:35:03 +01:00
Vincent Penquerc'h
7618699ffd audiobasesink: avoid possible sample count overflow 
At 48 kHz, 2<<31 samples is reached before 13 hours so it
sounds plausible this would be hit.

Coverity 1139800, 1139801
 2014-04-10 11:06:00 +01:00
Josep Torra
6ce7ade7c6 audioringbuffer: parse channels field from compressed audio caps 
Also parse channels as an optional field in the caps for compressed
audio formats.
 2014-04-08 12:54:04 +02:00
Vincent Penquerc'h
169166d0a2 audiobasesink: clip start samples to match clipped start time 
Clock slaving can clip start time to zero, giving us a shorted
duration than we originally got. To keep in sync, we must then
discard the samples falling before that zero timestamp.

This possibly fixes random distortion caused by constant PA
underflows which are never resynced.
 2014-04-04 17:04:06 +01:00
Rafał Mużyło
5496d09eb4 audio: map channels=1,channel-mask=0 to MONO instead of NONE 
Fixes problem in audioconvert, which would end up using
a mixmatrix when converting between different mono format
because it thinks MONO positioning is different from
unpositioned channels, which is not the case in this
special case. The mixmatrix would end up being 0.0 so
audioconvert would convert to silence samples.

https://bugzilla.gnome.org/show_bug.cgi?id=724509
 2014-02-18 10:41:47 +00:00
Sebastian Dröge
bc92cd8f67 audiosrc: Fix typo in docs 
We read *from* the audio device, not to it.
 2014-02-09 11:28:48 +01:00
Stefan Sauer
76ec6d3760 docs: doc fixes for audio library 
Add sections docs for audiometa. Fix sections docs for audiochannels. Remove old
mixerutil section.
 2014-02-03 09:36:43 +01:00
Thiago Santos
e00dc5b879 audioencoder: push pending events and tags before EOS 
if there are tags or events pending and an EOS is received, push those
events and tags before the EOS.
 2014-01-29 12:33:59 -03:00
Wim Taymans
6a88d6f8cd audiobasesink: make _get_time more threadsafe 
We call the _get_time function from the provided clock and we don't lock
the sink object for performance reasons. Make sure we only read and
check variables once so that they don't change while we are executing
the code.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720661
 2014-01-21 11:25:18 +01:00
Thiago Santos
695ddbd56f audiodecoder: copy rate and channels from input before fixating output caps 
For default caps generation when handling gap events that are sent
before any buffer, try to use caps that are closer to what upstream
provided to avoid fixating rate or channels to 1 as default.

So there are the steps:
1) Try to set rate, channels and channel-mask from upstream if provided
2) Fixate the rate and channels to the default rate and channels from
   audio lib
3) Fixate the caps just to be sure everything is fixed
4) If no channel-mask was provided and channels > 2, use a default
   channel-mask (taken from audioconvert code)

https://bugzilla.gnome.org/show_bug.cgi?id=722144
 2014-01-15 15:20:39 -03:00
Thiago Santos
95a56dbda7 audiodecoder: avoid parsing caps event if it is not used 
Saves some cpu
 2014-01-14 09:34:44 -03:00
Thiago Santos
8cf8332b91 audiodecoder: make sure caps is set before forwarding gap event 
Before trying to generate a default fixated caps when handling a gap
event, make sure that the same strategy that is used when handling
a buffer has been attempted. Otherwise audiodecoder will ignore
upstream caps settings such as rate and channels and will likely
end with a caps with channels=1 and rate=1.

https://bugzilla.gnome.org/show_bug.cgi?id=722144
 2014-01-14 09:34:44 -03:00
Jan Schmidt
f0b655e1ad audiobasesrc: Avoid unnecessary configuration 
Port a change from audiobasesink from def07410, to ignore setcaps
when the caps don't actually change, and avoid a reconfiguration
and reset of the ringbuffer in that case.
 2014-01-03 02:20:39 +11:00
Sebastian Dröge
58592a2af3 audio/video-info: Properly initialize the info structures in set_format() 
And don't assume in other code that set_format() preserves any fields at
all. These assumptions were already made here for fields that were changed
by set_format().
 2013-12-30 10:53:24 +01:00
Sebastian Dröge
65732d9c97 audio/video-info: Initialize the complete struct to 0 in the beginning 
Instead of only initializing some parts in some code paths. Also
makes it easier to use the reserved bits of the structs later.

https://bugzilla.gnome.org/show_bug.cgi?id=720810
 2013-12-30 10:15:20 +01:00
Reynaldo H. Verdejo Pinochet
5f07c1ed4e audiobasesrc: Bunch of cosmetic/grammar fixes 2013-12-27 01:36:09 -03:00
Reynaldo H. Verdejo Pinochet
0a6d6e1fff audiobasesrc: Retarget FIXME to 2.0 
Properly fixing this one would break API.
 2013-12-27 01:36:09 -03:00
Reynaldo H. Verdejo Pinochet
aa1883d5d7 audiobase*: Drop trailing withespaces 2013-12-27 01:36:09 -03:00
Reynaldo H. Verdejo Pinochet
d1b3454299 audiobasesrc: Break some too long lines 2013-12-27 01:36:09 -03:00
Reynaldo H. Verdejo Pinochet
6b17d86692 audiobasesrc: Add FIXME for times in NSECONDS 
Timebase is in nanoseconds pretty much everywhere else
 2013-12-27 01:36:09 -03:00
Jan Schmidt
c24a1254c9 audiodecoder: Choose a default initial caps before sending GAP 
If there are no caps from the audio decoder when handling a GAP
event - as when one is received right at the start on a DVD without
initial audio - then choose any default caps for downstream and
then send the GAP, so the audio sink has a configured format in
which to start the ringbuffer.

Also, make the audio sink reject a GAP without caps with a clearer
error message.

Fixes bug https://bugzilla.gnome.org/show_bug.cgi?id=603921
 2013-12-27 04:04:45 +11:00
Reynaldo H. Verdejo Pinochet
21190b9749 gstaudiobasesink: Always reset last_align 
Should be done for all the reset_sync() cases. Not
only for the READY to PAUSED one.
 2013-12-20 18:06:25 -03:00
Reynaldo H. Verdejo Pinochet
032779ff13 gstaudiobasesink: Reset last_align to 0, not -1 
This is the expected behavior in READY -> PAUSED
 2013-12-20 18:02:42 -03:00
Reynaldo H. Verdejo Pinochet
c1de7cdefb gstaudiobasesink: Always reset avg_skew on _reset 
Only case in which it wasn't (READY to PAUSED) should
have had this value reseted too.
 2013-12-20 17:58:43 -03:00
Reynaldo H. Verdejo Pinochet
adf800087c gstaudiobasesink: Retarget FIXME to 2.0 
Properly fixing this one would break API
 2013-12-20 17:48:22 -03:00
Reynaldo H. Verdejo Pinochet
d35db35258 gstaudiobasesink: Factor out reset sync routine 2013-12-20 17:47:38 -03:00
Reynaldo H. Verdejo Pinochet
b324d67586 gstaudiobasesink: Drop dead _sink_async_play() code 2013-12-20 13:58:34 -03:00
Reynaldo H. Verdejo Pinochet
2f04733a4b gstaudiobasesink: Break some too long lines 2013-12-20 13:58:33 -03:00
Reynaldo H. Verdejo Pinochet
187b106202 gstaudiobasesink: Cosmetics, grammar/spelling 
- Drop repeated 'yet' from debug msg
- Drop repeated 'to' from param desc
- Some spelling
 2013-12-20 13:58:33 -03:00
Edward Hervey
b97c711def audio/video: Initialize all {audio|video}info fields 
Fixes "Unitialized Scalar Variable" issues reported by Coverity.

Has the added advantage of detecting whether somebody *does* use those
fields (ending up with a invalid address).

https://bugzilla.gnome.org/show_bug.cgi?id=720810
 2013-12-20 14:47:22 +01:00
Reynaldo H. Verdejo Pinochet
86b0a0d6d0 gstaudiobasesink: Refactor alignment computation for clarity 2013-12-19 18:05:44 -03:00
Todd Agulnick
38d8fa12a5 Some compiler warning fixes to satisfy XCode compiler 
https://bugzilla.gnome.org/show_bug.cgi?id=720513
 2013-12-16 16:51:29 +01:00
Wim Taymans
df3718ea2b audiobasesink: handle the RESYNC flag 
Also resync when a buffer with the RESYNC flag is seen.
 2013-12-05 16:27:35 +01:00
Julien Isorce
e68317f070 audiodec/enc: clear reconfigure flag if negotiate succeeds 
So that it avoids to send an allocation query twice.
One from an early call to gst_audio_encoder_negotiate from a
subclass, then one from gst_audio_encoder_allocate_output_buffer.
Which means that previously gst_audio_encoder_negotiate was not
clearing the GST_PAD_FLAG_NEED_RECONFIGURE even on success.

Fixes bug https://bugzilla.gnome.org/show_bug.cgi?id=719684
 2013-12-05 15:19:16 +00:00
Sebastian Dröge
400d4baf92 audiodecoder: Use FALSE instead of 0 2013-12-05 11:37:09 +01:00
Mark Nauwelaerts
6e639b73ff audiodecoder: no fallback to segment start for reverse playback 
See https://bugzilla.gnome.org/show_bug.cgi?id=709965
 2013-12-04 19:24:25 +01:00
Mark Nauwelaerts
387e5f0c14 audiodecoder: use segment start as fallback ts if no other available 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=709965
 2013-12-02 20:36:21 +01:00
Sebastian Dröge
f8477e6b88 audiodecoder: error out if no frames are decoded before eos 
Raise an error in case no frames are decoded before EOS and we
have input, meaning that data was received but it was somehow invalid.

Based on the videodecoder change, merged here for consistency.

https://bugzilla.gnome.org/show_bug.cgi?id=711094
 2013-11-26 12:29:30 +01:00
Sebastian Dröge
b0788ce054 audiodecoder: Allow using -1 for infinite tolerated errors 
Allows using -1 to make audiodecoder never post an error message
after decoding errors.

Based on the videodecoder change, merged here for consistency.

https://bugzilla.gnome.org/show_bug.cgi?id=711094
 2013-11-26 12:20:33 +01:00
Mark Nauwelaerts
b13a722746 audioencoder: also set output buffer DTS 2013-11-16 15:25:38 +01:00
Sebastian Dröge
3fb235c53c audio: Update ORC dist files 2013-11-03 15:58:35 +01:00
Sebastian Dröge
081f009e25 audio-format: Use ORC for filling memory with silence samples 2013-11-03 15:58:35 +01:00
Takashi Iwai
6d659e3c6f audioringbuffer: Don't clear need_reorder flag too early 
gst_audio_ring_buffer_set_channel_positions() checks whether the given
positions are identical with the current setup and returns
immediately if so.  But it also clears need_reorder flag before this
comparison, thus this flag might be wrongly cleared if the function is
called twice with the same channel positions.

Move the flag clearance after the check.

https://bugzilla.gnome.org/show_bug.cgi?id=709754
 2013-10-09 19:00:33 +02:00
Johannes Dewender
019ef0747d audiocdsrc: Don't consider trailing data tracks for MusicBrainz disc id calculation 
MusicBrainz removes trailing data tracks from releases on the server
and also for the calculation of the MusicBrainz Disc ID.

https://bugzilla.gnome.org/show_bug.cgi?id=708991
 2013-10-01 22:24:22 +02:00
David Svensson Fors
09d628f8f1 audioringbuffer: check if acquired in set_timestamp 
Also use GST_OBJECT_LOCK when accessing object data in set_timestamp.

https://bugzilla.gnome.org/show_bug.cgi?id=702230
 2013-10-01 22:12:07 +02:00
Matej Knopp
dbaf1bf0a3 audio: change buffer timestamp when clipping even if data hasn't been trimmed 
https://bugzilla.gnome.org/show_bug.cgi?id=708952
 2013-09-28 11:39:43 +02:00
Wim Taymans
c9ff3e4f98 audiobasesink: do big correction for large drift 
If we are using skew slaving and we drift more than twice the allowed amount, do
a big correction to get back on track more quickly.
 2013-09-25 16:03:07 +02:00
Sebastian Dröge
420e229829 audioencoder/decoder: Mark pads as requiring reconfiguration again if negotiation fails 
Otherwise we might end up in non-optimal configuration, especially
when a flush happened during reconfiguration.
 2013-09-12 09:42:36 +02:00
Wim Taymans
d3641943b3 docs: fix some doc blocks 2013-09-09 15:52:05 +02:00
Mathieu Duponchelle
d1cb9c994b video/audio: #define metadata strings. 
For instance "orientation" becomes GST_VIDEO_ORIENTATION_METADATA.
 2013-09-09 15:37:02 +02:00
Sebastian Dröge
96ab6db422 audioencoder: Simplify pushing of pending events during negotiation 
And also don't send the same caps twice.
 2013-08-23 19:17:16 +02:00
Sebastian Dröge
daf017ced8 audiodecoder: Fix last commit and simplify code a lot 2013-08-23 19:10:48 +02:00
Edward Hervey
f9ebfd57f8 audiodecoder: Fix previous commit 
(sorry)
 2013-08-23 16:59:30 +02:00
Edward Hervey
cd3fe60c68 audiodecoder: Don't push out identical caps 
This avoids triggering plenty of extra code/methods/overhead downstream when
we can just quickly check whenever we want to set caps whether they are
identical or not

https://bugzilla.gnome.org/show_bug.cgi?id=706600
 2013-08-23 15:22:05 +02:00
Tim-Philipp Müller
6b070784c4 audio: make direct includes work again 
Not nice to break people's code if we can avoid it. Could
add a warning in the next cycle, and then require single
includes in the cycle after.

https://bugzilla.gnome.org/show_bug.cgi?id=695889
 2013-08-16 14:14:11 +01:00
Youness Alaoui
ca2a515373 audiodecoder: Clear taglist on reception of a STREAM_START event 
https://bugzilla.gnome.org/show_bug.cgi?id=705109
 2013-08-12 13:02:59 +02:00
Matej Knopp
197376212c audiodecoder: do not leak input caps 
https://bugzilla.gnome.org/show_bug.cgi?id=704926
 2013-07-26 15:37:04 +01:00
Sebastian Dröge
99ef452fc4 audio/videodecoder: Rename variable in macro from dec to __dec 
Otherwise it might shadow another variable in the outside scope
and cause interesting side effects.
 2013-07-25 14:11:28 +02:00
Sebastian Dröge
50fd867a43 audioencoder: Don't return not-negotiated if flushing 
If the pad is flushing after a failed negotiation, return
GST_FLOW_FLUSHING instead from finish_frame().

https://bugzilla.gnome.org/show_bug.cgi?id=701763
 2013-06-30 18:17:42 +02:00
Mathieu Duponchelle
97e68b36c7 audiodecoder: Don't return not-negotiated if flushing 
If the pad is flushing after a failed negotiation, return GST_FLOW_FLUSHING.

https://bugzilla.gnome.org/show_bug.cgi?id=701763
 2013-06-25 12:51:55 -04:00
Jonas Holmberg
82e5ec553b audioencoder: unref before memset 
Unref allocator and input_caps in encoder context before memsetting the
context.
 2013-06-19 13:56:28 +02:00
Ognyan Tonchev
f240d34c7e audiobasesrc: add 2 missing gst_buffer_unmap () calls 
There are 2 missing calls to gst_buffer_unmap () in the error handling in
create ().

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702467
 2013-06-17 16:34:26 +02:00
Sebastian Dröge
ff5d3313d4 Release 1.1.1 2013-06-05 18:31:27 +02:00
Sebastian Dröge
c06377b385 audioencoder: Remove private copy of gst_audio_info_is_equal() 
And improve the public one a bit based on it.
 2013-06-01 09:06:22 +02:00
Sebastian Dröge
5065e76b1c audio: Add gst_audio_info_is_equal() 2013-05-30 23:56:52 +02:00
Sebastian Dröge
b8c6413a8e audio: Always provide a buffer in gst_audio_(enc|dec)oder_allocate_output_buffer() 
We have no way of tell the caller of the exact error (e.g. if we're flushing),
so will have to wait until the caller uses API that returns a GstFlowReturn,
for example when pushing this buffer.

https://bugzilla.gnome.org/show_bug.cgi?id=700006
 2013-05-24 16:54:46 +02:00
Alexander Schrab
a049b102da alsasrc: Make using driver timestamps possible 
https://bugzilla.gnome.org/show_bug.cgi?id=699744
 2013-05-20 11:25:17 +02:00
Sebastian Dröge
be154ee9d6 audio-info: Always pass NULL as position parameter to gst_audio_info_set_format() 
https://bugzilla.gnome.org/show_bug.cgi?id=700259
 2013-05-15 09:26:56 +02:00
Sebastian Dröge
b401f447d2 audio-info: For more than 64 channels don't allow a channel layout 
More than 64 channels have all channels unpositioned.

https://bugzilla.gnome.org/show_bug.cgi?id=700259
 2013-05-14 09:34:21 +02:00
Sebastian Dröge
351405d8a0 audio: Make sure to push pre-caps events before the caps event 2013-05-08 15:56:34 +02:00
Tim-Philipp Müller
f5c0d61be7 Update disted orc backup files 
Generated with 0.4.17 now.
 2013-04-22 13:58:33 +01:00
Sebastian Dröge
d537a21075 audioencoder: Ignore caps events if the input caps did not change 2013-04-18 09:58:36 +02:00
Sebastian Dröge
d1a08af605 audiodecoder: Ignore caps events if the input caps did not change 2013-04-18 09:58:36 +02:00
Tim-Philipp Müller
e96ca66c36 docs: add some more audio macros 2013-04-17 09:26:40 +01:00
Sebastian Dröge
98f41f1c39 audioringbuffer: Also reset segbase 2013-04-15 10:13:14 +02:00
Paul HENRYS
587b2721c8 audioringbuffer: Reset segdone when releasing audioringbuffer 
https://bugzilla.gnome.org/show_bug.cgi?id=697723
 2013-04-15 10:09:49 +02:00
Wim Taymans
76d71da1c4 audiodecoder: don't make negative timestamp 
Clamp timestamp interpollation to 0 to avoid going negative. This should not
happen, really, but until the interpolation is improved this seems better.
 2013-03-31 13:46:30 +02:00
Wim Taymans
03f658dda2 audiodecoder: forward stream-start immediately 2013-03-30 19:14:37 +01:00
Stefan Sauer
e4ee1dde02 audioencoder: api doc fixes. 2013-03-29 10:33:35 +01:00
Paul HENRYS
78a8531c75 audiobasesrc: Fix ringbuffer handling when settings caps 
ringbuffer was released after setting values to its spec field
in gst_audio_base_src_setcaps(). This led to failure in case
gst_audio_base_src_setcaps() is called more than one time.

https://bugzilla.gnome.org/show_bug.cgi?id=696540
 2013-03-25 10:16:03 +01:00
Marc Leeman
0fa50b44f0 audioringbuffer: avoid division by 0 when outputting debug info 
https://bugzilla.gnome.org/show_bug.cgi?id=695832
 2013-03-15 09:06:07 +00:00
Akihiro Tsukada
a32877125f audio: add support for AAC pass-through 
https://bugzilla.gnome.org/show_bug.cgi?id=694443
 2013-02-27 00:38:05 +00:00
Stefan Sauer
b274ff7c21 audioringbuffer: log a few more details (e.g. obj-name) 2013-02-25 19:55:00 +01:00
Tim-Philipp Müller
6682215d9d audio: fix GST_AUDIO_INFO_ENDIANNESS macro 2013-02-16 13:06:54 +00:00
Tim-Philipp Müller
664adc6e19 gst-libs: use GST_*_1_0 environment variables everywhere 
The _1_0 suffixed environment variables override the
non-suffixed ones, so if we're in an environment that
sets the _1_0 suffixed ones, such as jhbuild, we need
to set those to make sure ours actually always get
used.
 2013-01-16 10:16:27 +00:00
Tim-Philipp Müller
b4def63f55 audio: don't use uninitialized variable in debug log 
https://bugzilla.gnome.org/show_bug.cgi?id=667317
 2012-12-29 14:29:53 +00:00
Wim Taymans
fe93457191 audioclock: mark as using some other clock 
We need to mark our clock as using some other clock source. Alsa source uses the
clock type to decide if it can use alsa driver timestamps or not.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=690465
 2012-12-20 16:48:04 +01:00
Wim Taymans
5e04fcd2ef audiobasesrc: init variable 
We need to initialize this variable because we can't be sure that the subclass
will set it.
 2012-12-20 16:47:56 +01:00
Tim-Philipp Müller
68f366a8d3 audiobasesrc: bail out if subclass posts an error 
Use new ringbuffer ERROR state to make all the various
threads bail out correctly when the subclass posts an
error. It's a bit iffy to communicate this properly
between the different bits of code.

https://bugzilla.gnome.org/show_bug.cgi?id=690197
 2012-12-17 20:50:32 +00:00
Tim-Philipp Müller
4f49c7a33b audioringbuffer: add GST_AUDIO_RING_BUFFER_STATE_ERROR state 
API: GST_AUDIO_RING_BUFFER_STATE_ERROR

https://bugzilla.gnome.org/show_bug.cgi?id=690197
 2012-12-17 20:50:32 +00:00
Thiago Santos
929edc2572 audiobasesrc: Always resync the ringbuffer on the first buffer 
In SKEW mode, use next_sample == -1 to check for the first sample
when starting to read samples so it resyncs the ringbuffer and
timestamps are ok.

Suggestion from Teemu Katajisto <teemu.katajisto@digia.com>

https://bugzilla.gnome.org/show_bug.cgi?id=648359
 2012-12-17 11:47:34 +01:00
Sebastian Dröge
3f82e919dd libs: Use foo/foo.h as single-include header consistently everywhere 
https://bugzilla.gnome.org/show_bug.cgi?id=688785
 2012-12-12 17:13:10 +00:00
Tim-Philipp Müller
fbff6c6fb1 audioencoder: add some more debug info and remove obsolete comment 2012-12-02 12:33:43 +00:00
Tim-Philipp Müller
8827437b61 audio: remove bogus Since marker from docs 
It was causing perl warnings in gtk-doc code.
 2012-11-21 23:19:14 +00:00
Evan Nemerson
4d77fba46c libs: Add missing single include headers and use them in GIRs 2012-11-21 11:01:24 +01:00
Tim-Philipp Müller
71e46b2478 gst_adapter_prev_timestamp -> gst_adapter_prev_pts 
https://bugzilla.gnome.org/show_bug.cgi?id=675598
 2012-11-14 00:03:15 +00:00
Sebastian Dröge
32139f9a3d audio: Use new GType for GThread instead of just G_TYPE_POINTER 2012-11-12 11:45:47 +01:00
Sebastian Dröge
d209727644 audiodecoder: Reset error count to 0 after successfully decoding a frame 2012-11-09 16:48:54 +01:00
Tim-Philipp Müller
5f59b4f7ee Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
 2012-11-03 23:05:09 +00:00
Miguel Angel Cabrera Moya
244fdcc69a audiobasesink: use the same type as the internal type to return it 
https://bugzilla.gnome.org/show_bug.cgi?id=687466
 2012-11-02 19:52:38 +00:00
Wim Taymans
5f44303925 audioringbuffer: reset spec on _release 
Reset the caps and the audioinfo when releasing the ringbuffer.
Fixed a bug with reusing pulsesink.
 2012-10-30 10:33:04 +00:00
Tim-Philipp Müller
a4f2df6341 Revert "g-i: change g-ir-scanner arg --library=libgstfoo-X.la to --library=gstfoo-X" 
This reverts commit e39fbe6b7e.

Looks like we need to pass the full .la file after all in a setup
with libtool, or it might not find the library, e.g. like

  ERROR: can't resolve libraries to shared libraries: gstfft-1.0

Conflicts:
	gst-libs/gst/audio/Makefile.am
	gst-libs/gst/pbutils/Makefile.am

Also see https://bugzilla.gnome.org/show_bug.cgi?id=603710
 2012-10-29 12:47:05 +00:00
Tim-Philipp Müller
973f4f09ea audio: try harder to make g-i use the build-tree libgsttag 
without adding additional --library= tags, which shouldn't be there.

https://bugzilla.gnome.org/show_bug.cgi?id=679315
 2012-10-28 17:59:27 +00:00
Tim-Philipp Müller
e39fbe6b7e g-i: change g-ir-scanner arg --library=libgstfoo-X.la to --library=gstfoo-X 
As it should be according to the man page.

https://bugzilla.gnome.org/show_bug.cgi?id=679315
 2012-10-28 17:35:57 +00:00
Mark Nauwelaerts
45d802b63f audiodecoder: track forced decoding state 2012-10-24 14:46:22 +02:00
Sebastian Dröge
1813701ef2 audiobasesink: Add explanation to the GAP event handling code 2012-10-24 11:22:29 +02:00
Sebastian Dröge
b793d0bfae audiobasesink: Properly handle GAP events 
These are now converted into silence buffers if they have
a duration or cause the ringbuffer and clock to be started
if they don't have a duration.

Fixes bug #685273.
 2012-10-24 11:19:05 +02:00
Tim-Philipp Müller
277ca04976 audiodecoder: don't leak message strings when error is not fatal 
https://bugzilla.gnome.org/show_bug.cgi?id=681192
 2012-10-20 11:38:10 +01:00
Tim-Philipp Müller
3ee2ad255b audiocdsrc: mention TOCs in docs 2012-10-17 19:59:57 +01:00
Mark Nauwelaerts
162433795a audio: properly handle clipping of empty buffer 2012-10-15 18:48:01 +02:00
Josep Torra
d8d9f0db97 audiodecoder: set of base_ts for segment formats other than time 
Fixes setting of converted segment start as base_ts when estimate rate
is allowed.
 2012-10-11 13:17:01 +02:00
Sebastian Dröge
e779002cfd audiodecoder: Don't unref caps twice 
Thanks to Josep Torra for noticing.
 2012-10-10 15:50:31 +02:00
Wim Taymans
3591df23b1 docs: playbin2 -> playbin 2012-10-09 12:20:10 +02:00
Andoni Morales Alastruey
8a5cf5ef4d audio/video: update documentation for vfunc's that require chaining up 2012-10-08 13:04:02 +02:00
Tim-Philipp Müller
cdb22274e6 audioencoder: make stop() vfunc also optional 
Just change default value, since we also don't want to fail
if we want to deactivate and aren't active or want to activate
and are already active.

https://bugzilla.gnome.org/show_bug.cgi?id=685490
 2012-10-04 13:40:32 +01:00
Andoni Morales Alastruey
795d366a0c audioencoder: don't fail if the start vfunc is not implemented 
Fix behaviour to match documentation and decoder class behaviour.

https://bugzilla.gnome.org/show_bug.cgi?id=685490
 2012-10-04 13:14:10 +01:00
Michael Smith
92560517e8 Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base 2012-10-03 10:45:26 -07:00
Michael Smith
a29c4f9489 meta registration: use g_once functions to register these threadsafely. 2012-10-03 10:44:59 -07:00
Arun Raghavan
9f9718715a audio: Explicitly specify endianness for IEC 61937 payloading 
This is required since some systems (DirectSound and OS X) manage the
final byte order themselves.

https://bugzilla.gnome.org/show_bug.cgi?id=678021
 2012-09-19 09:15:16 +05:30
Mark Nauwelaerts
c629a44162 replace gst_tag_list_free with gst_tag_list_unref 2012-09-14 17:53:21 +02:00
Wim Taymans
a57198a0ba audio: improve property description 
Improve the description of the latency-time and buffer-time properties in the
audio sink and source.
 2012-09-14 16:08:50 +02:00
Sebastian Dröge
6e33f2d464 audiodecoder: Don't output an (unreffed) buffer in error cases 2012-09-14 14:54:22 +02:00
Tim-Philipp Müller
f7c6aa5abd Release 0.11.94 2012-09-14 02:47:54 +01:00
Olivier Crête
b35bc51ed6 audio: Fix annotations 2012-09-13 17:11:56 -04:00
Wim Taymans
0ce33461c8 audiosrc: check for flushing state in provide_clock 
Only provide a clock when we are not flushing, this means that we have posted a
PROVIDE_CLOCK message. We used to check if we were acquired but that doesn't
work anymore now that we do the negotiation async in the streaming thread: it's
possible that we are still negotiating when the pipeline asks us for a clock.
 2012-09-10 12:19:22 +02:00
Wim Taymans
44dab50b7a ringbuffer: add method to check the flushing state 2012-09-10 12:19:22 +02:00
Mark Nauwelaerts
75fe950c33 gst-libs: restore original full padding 2012-09-10 11:45:44 +02:00
Pontus Oldberg
a2f8ec4f5a ringbuffer: add support for timestamps 
Make it possible for subclasses to provide the timestamp (as an absolute time
against the pipeline clock) of the last read data.
Fix up alsa to provide the timestamp received from alsa. Because the alsa
timestamps are in monotonic time, we can only do this when the monotonic clock
has been selected as the pipeline clock.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=635256
 2012-09-10 11:34:14 +02:00
Mark Nauwelaerts
a29fab200c audio{de,en}coder: use GstClockTime parameters where appropriate 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=683672
 2012-09-10 11:20:50 +02:00
Thibault Saunier
dc5bb008a3 audio: port to the new GLib thread API 2012-09-09 20:41:06 -03:00
Tim-Philipp Müller
2079a8c12b Remove glib-compat-private.h stuff we don't need any more 
It's all been ported to the latest GLib API now.
 2012-09-09 18:36:49 +01:00
Mark Nauwelaerts
c9d3f32cc9 audioencoder: plug some leaks 2012-09-06 12:16:59 +02:00
Wim Taymans
668ce33384 update for basesink change 2012-09-04 12:18:11 +02:00
Tim-Philipp Müller
a99a1042b9 gst_message_new_duration() -> gst_message_new_duration_changed() 2012-09-02 01:27:17 +01:00
Jan Schmidt
5dafecad31 audiodecoder: Handle GAP events in place of segment updates 
Use them to trigger generation of an empty output buffer or
to send pending events downstream and trigger pre-roll
 2012-08-31 12:42:12 -07:00
Edward Hervey
def07410ef audiobasesink: Avoid resetting ringbuffer when not needed 
If the ringbuffer was configured to the same caps as previously, we
don't need to reconfigure it.
 2012-08-14 18:56:00 +02:00
Víctor Manuel Jáquez Leal
f7f0c55e5f audiodecoder: getter for allocator 
Sometimes the decoder would use the allocator for something else than just
allocating output buffers, for example, querying different parameters.

This patch expose a getter accessor for the negotiated memory allocator.
 2012-08-14 15:47:34 +02:00
Víctor Manuel Jáquez Leal
936ec3eb8f audioencoder: getter for allocator 
Sometimes the encoder would use the allocator for something else than just
allocating output buffers, for example, querying different parameters.

This patch expose a getter accessor for the negotiated memory allocator.
 2012-08-14 15:47:29 +02:00
Tim-Philipp Müller
2ff4d2efe3 audioencoder: return TRUE from _set_output_format() if all is good 
Fixes not-negotiated errors in wavpackenc unit test.
 2012-08-13 23:34:52 +01:00
Sebastian Dröge
62ec7f837d audioencoder: Let global tag events be handled the same way as other events 2012-08-09 17:06:31 +02:00
Sebastian Dröge
e9fbba63b5 audiodecoder: Let global tag events be handled the same way as other events 2012-08-09 16:55:19 +02:00
Sebastian Dröge
2a1f8a4da3 audio: Merge upstream stream tags 2012-08-09 16:24:47 +02:00
Sebastian Dröge
7f0e65bb46 audio: Always keep a complete taglist around 
Otherwise updates to the tags will cause non-updated
tags to be lost downstream.
 2012-08-09 15:48:03 +02:00
Sebastian Dröge
bc4d923982 audioencoder: Add negotiate vfunc that is used to negotiate with downstream 
The default implementation negotiates a buffer pool and allocator
with downstream.
 2012-08-09 15:27:33 +02:00
Sebastian Dröge
9309272309 audioencoder: Decouple setting of output format and downstream negotiation 
This makes the audio encoder base class more similar to the video
encoder base class.
 2012-08-09 15:21:01 +02:00
Sebastian Dröge
513d4f7cd1 audiodecoder: Add negotiate vfunc that is used to negotiate with downstream 
The default implementation negotiates a buffer pool and allocator
with downstream.
 2012-08-09 15:10:05 +02:00
Sebastian Dröge
e1702d62a0 audiodecoder: Decouple setting of output format and downstream negotiation 
This makes the audio decoder base class more similar to the video
decoder base class.
 2012-08-09 15:02:27 +02:00
Tim-Philipp Müller
6422f2d085 Update .gitignore 2012-08-08 09:06:30 +01:00
Tim-Philipp Müller
ca31913c04 audiocdsrc: update for TOC API change 2012-07-28 11:13:12 +01:00
Sebastian Dröge
99d73c94e9 tag: Update for taglist/tag event API changes 2012-07-28 00:35:02 +02:00
Wim Taymans
683a38ad65 update for new variable names 2012-07-27 15:24:43 +02:00
Wim Taymans
40a0624e99 audio-format: fix shift for 18 bits samples 
The 18bits of the sample are in the LSB so we need to shift them 14 positions to
bring them to 32 bits.
 2012-07-26 15:42:38 +02:00
Mark Nauwelaerts
c91615bd82 audio{de,en}coder: delay input caps processing until processing data 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680614
 2012-07-26 14:35:30 +02:00
Mark Nauwelaerts
28537dc73c audioencoder: avoid setting output caps twice 
... which may not be handled or appreciated well downstream,
e.g. muxers only performing header setup once.
 2012-07-25 15:58:19 +02:00
Mark Nauwelaerts
1f962bc108 audioencoder: also consider filter caps in getcaps 2012-07-25 15:58:19 +02:00
Mark Nauwelaerts
26d74941fb Revert "audioencoder: plug caps ref leak" 
This reverts commit 08ff5899a7.

Was not a leak to begin with as we did not have ownership of caps.
 2012-07-25 12:30:54 +02:00
Mark Nauwelaerts
08ff5899a7 audioencoder: plug caps ref leak 2012-07-25 11:58:26 +02:00
Mark Nauwelaerts
473371f943 audiodecoder: hold caps ref while needed 2012-07-25 11:58:26 +02:00
Mark Nauwelaerts
d55529621c audioencoder: correctly compare audio info positions 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680553
 2012-07-25 11:58:26 +02:00
Mark Nauwelaerts
65ea6dee60 audiodecoder: only arrange to reconfigure if data provided 
... otherwise audio format need not be known already.
 2012-07-24 14:48:59 +02:00
Mark Nauwelaerts
d63a4024b8 audiodecoder: minor doc fix 2012-07-24 12:30:21 +02:00
Wim Taymans
5ff002b47a audio: prefix orc_* functions with audio_orc_* 
To avoid potential conflicts in other modules when statically linking
 2012-07-23 17:16:34 +02:00
Sebastian Dröge
d55d7fdc38 audio: Renegotiate if necessary 
And also correct usage of the base class stream lock.
 2012-07-23 12:01:12 +02:00
Sebastian Dröge
7b06c34868 audiodecoder: Handle allocation query 2012-07-23 11:42:22 +02:00
Sebastian Dröge
0814d38e98 audiodecoder: Add propose_allocation, decide_allocation vfuncs and functions to allocate buffers with information from the allocation query results 2012-07-23 10:28:05 +02:00
Sebastian Dröge
0513d3d9f4 audioencoder: Add propose_allocation, decide_allocation vfuncs and functions to allocate buffers with information from the allocation query results 2012-07-23 10:20:05 +02:00
Edward Hervey
55f692eff6 audiodecoder: Don't assert on pad caps not being set 
The decoder might have been de-activated in the meantime (resulting
in NULL pad caps).

If the decoder really isn't configured, then it will error out further
down when checking whether the GST_AUDIO_INFO_IS_VALID()

https://bugzilla.gnome.org/show_bug.cgi?id=667562
 2012-07-19 10:55:53 +02:00
Evan Nemerson
7a7374f2ef audiometa: add missing array array annotations 2012-07-17 11:07:18 +02:00
Evan Nemerson
17815020fd audio: add missing array and element-type annotations for binary data 2012-07-17 11:06:57 +02:00
Evan Nemerson
fd91104636 audio-channels: add missing array-related annotations 2012-07-17 11:06:47 +02:00
Evan Nemerson
1606028c08 audioencoder: add missing element-type to set_headers method 2012-07-17 11:06:22 +02:00
Edward Hervey
2817bdadc9 libs: Remove "Since" markers and minor doc fixups 2012-07-13 12:11:06 +02:00
Edward Hervey
c9428c96b1 baseaudiosink: Resync when ringbuffer resets 
When the ringbuffer gets restarted (like in setcaps), we *will* have
to resync against the new values.

Without this we end up blindly assuming the new samples align to the
old ones.
 2012-07-12 09:51:35 +02:00
Sebastian Dröge
9de1b170b3 audiocdsrc: Remove the TOC query handling 2012-07-05 12:35:35 +02:00
Sebastian Dröge
0ac1596d8d audiocdsrc: Update for TOC API changes 2012-07-05 12:29:00 +02:00
Sebastian Dröge
b362ec3a57 audiocdsrc: Only push TOC event, the TOC message is handled by the sinks 2012-07-03 17:31:54 +02:00
Tim-Philipp Müller
df70b2d2ce audiocdsrc: send TOC event downstream if we're in continuous mode 
If we're in continuous mode where we'll play the entire CD from
start to finish, send a TOC event downstream so any downstream
muxers can write a TOC to indicate where the various tracks
start and end.
 2012-06-28 23:41:16 +01:00
Tim-Philipp Müller
b27c649a48 audiocdsrc: post TOC message on the bus on start-up 
First attempt at implement the various GstToc API
bits in GstAudioCdSrc.

https://bugzilla.gnome.org/show_bug.cgi?id=668996
 2012-06-26 19:53:35 +01:00
Tim-Philipp Müller
a821d428bb audio: make sure g-i doesn't parse orc-generated gstaudiopack.h file 2012-06-24 00:28:40 +01:00
Wim Taymans
c003efcc63 audiobasesink: fix for basesink API change 2012-06-18 11:40:36 +02:00
Jan Schmidt
d9740bf9ba audio decoder: Add some debug output for bad caps from children 2012-06-12 23:52:35 +10:00
Vincent Penquerc'h
f8b8711081 audiodecoder: push queued events only when we have a first buffer 
https://bugzilla.gnome.org/show_bug.cgi?id=675812
 2012-06-11 11:29:13 +01:00
Wim Taymans
9d6967fe9a Add generated orc files 2012-06-08 17:57:43 +02:00
Wim Taymans
12ac9f0aa2 Also build the orc generated code 2012-06-08 17:57:43 +02:00
Wim Taymans
3f8c5ea036 audio: add orc enabled pack and unpack functions 2012-06-08 17:57:43 +02:00
Wim Taymans
8e393d898a audio: add flag to mark possible unpack formats 
Make a new flag to mark formats that can be used in pack and unpack functions.
Mark S32NE and F64NE as those unpack formats
 2012-06-08 17:57:43 +02:00
Sebastian Dröge
462c4cc3d8 audio: Remove unused, generated marshallers 2012-06-08 11:28:56 +02:00
Wim Taymans
3da0b71876 audio: split audio header into logical parts 2012-06-08 10:10:08 +02:00
Wim Taymans
a2172bdb4b update for tag event change 2012-06-06 13:05:47 +02:00
Sebastian Dröge
2667d4bb82 Revert "audiodecoder: Error out earlier in a few places if something goes wrong" 
This reverts commit eb68a2d5a7.

This sometimes errors out too early now, needs some more thoughts.
 2012-06-04 10:01:42 +02:00
Sebastian Dröge
f609b3a627 audiodecoder: Return setcaps return value instead of always TRUE 2012-06-04 09:56:30 +02:00
Sebastian Dröge
eb68a2d5a7 audiodecoder: Error out earlier in a few places if something goes wrong 2012-06-02 17:16:13 +02:00
Wim Taymans
c66da2c74b audio: add flags for the pack/unpack functions 
Add a flag argument to the pack and unpack function so that we can expand it
later when needed. We could for example prefer a High Quality pack/unpack
operation later.
 2012-05-29 09:54:43 +02:00
Arun Raghavan
9c29cd70ee audio: Fix DTS IEC61937 payloading 
DTS type I-III specify the burst length in bits. Only type IV (which we
do not currently support) needs it to be specified in bytes. Thanks to
Julien Moutte for pointing this out.
 2012-05-25 12:38:32 +02:00
Sebastian Rasmussen
b7b123964b gst-libs: make pkg-config get path to pkg-config dirs from configure 
When --with-pkg-config-path is supplied to configure this path is now
explicitly propagated to pkg-config.

https://bugzilla.gnome.org/show_bug.cgi?id=673377
 2012-05-05 23:26:20 +01:00
Sebastian Dröge
69b18ab09d gst-libs: Remove interfaces libs and mixer/tuner interfaces 
The navigation interface is now in the video library.
 2012-04-13 13:14:13 +02:00
Alban Browaeys
6c8abf24cf libs: Link against internal tag library 2012-04-11 09:58:49 +02:00
Sebastian Dröge
8091546694 audio: Remove obsolete FIXME 0.11 2012-04-11 09:57:35 +02:00
Alessandro Decina
ebf80977c4 audiodecoder: don't discard timestamps when consecutive input buffers have the same ts 
Avoid pushing out buffers with the same timestamp only if the out buffers are
decoded from the same input buffer. Instead keep the timestamps when upstream
pushes consecutive buffers with the same ts.
 2012-04-05 10:19:46 +02:00
Mark Nauwelaerts
6eeca397fc audioencoder: plug a definite and rare leak 2012-04-04 19:57:35 +02:00
Sebastian Dröge
65307dd132 gst: Update versioning 2012-04-04 14:55:15 +02:00
Mark Nauwelaerts
91aa1eb7dd audio{de,en}coder: fixup documentation 2012-04-02 14:23:33 +02:00
Sebastian Dröge
b701534204 audioencoder: Fix handling of offset/offset-end for Ogg codecs 
Fixes the vorbisenc unit test.
 2012-03-31 12:55:15 +02:00
Sebastian Dröge
a103fa85a9 audio{en,de}coder: Track input and output segments separately 
They can go out of sync for some time if processing of buffers
on the old segment happens after the segment was received.
 2012-03-30 13:21:09 +02:00
Sebastian Dröge
9cd9f00799 audioencoder: Add gst_audio_encoder_set_headers() to the docs 2012-03-30 12:57:02 +02:00
Sebastian Dröge
78bcb67ea5 audioencoder: Add function to set in-stream headers 
API: gst_audio_encoder_set_headers()

This makes the hack in vorbisenc and probably others in ::pre_push()
unnecessary.
 2012-03-30 12:47:28 +02:00
Sebastian Dröge
f791ec1f10 audioencoder: Rename ::event() to ::sink_event() and add ::src_event() 2012-03-30 12:23:13 +02:00
Sebastian Dröge
d8cb235fe4 audiodecoder: Rename ::event() to ::sink_event() and add ::src_event() 2012-03-30 12:23:13 +02:00
Sebastian Dröge
40a4f2f8aa audiodecoder: Rename _byte_time() to _estimate_rate() 
Which is telling more about what this actually does and is more
consistent with the video base classes.
 2012-03-30 11:51:47 +02:00
Mark Nauwelaerts
2ddc6bb63d audiodecoder: handle downstream seeking query 
... or not, in line with how segment events are treated.
 2012-03-28 16:41:01 +02:00
Wim Taymans
77a4f5865b audioencoder: avoid caps copy 2012-03-27 15:44:43 +02:00
Wim Taymans
32bd12dba9 Merge branch 'master' into 0.11 
Conflicts:
	.gitignore
	common
	configure.ac
	ext/vorbis/gstvorbisdeclib.h
	gst-libs/gst/audio/gstaudioencoder.c
	gst-libs/gst/riff/riff-read.c
	gst/playback/gstplaysink.c
	gst/playback/gstplaysinkconvertbin.c
	tests/check/libs/video.c
 2012-03-22 11:35:13 +01:00
Wim Taymans
a619d3a8b0 update for memory api changes 2012-03-20 13:20:36 +01:00
Mark Nauwelaerts
278b0f093b audio: include audio enumtypes 2012-03-19 16:18:56 +01:00
Wim Taymans
dfb8e7cb2c don't pass random pointers to pull_range 2012-03-16 21:46:47 +01:00
Wim Taymans
4e1ed6f649 audio: fix debug line 2012-03-13 12:39:52 +01:00
Wim Taymans
25137962ad fix for caps API changes 2012-03-11 19:04:41 +01:00
Wim Taymans
7296ef7c63 audiobasesink: add some G_LIKELY 2012-03-09 17:15:38 +01:00
Wim Taymans
94869bff38 audio: avoid buffer copy when nothing is clipped 
when nothing is clipped, return the input buffer instead of creating and
returning an identical copy.
 2012-03-09 16:17:54 +01:00
Sebastian Dröge
7ff608889a audio{en,de}coder: Add optional open/close vfuncs 
This can be used to do something in NULL->READY, like checking
if a hardware codec is actually available and to error out early.
 2012-03-09 10:56:07 +01:00
Tim-Philipp Müller
29c266ccff Merge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	common
	docs/libs/gst-plugins-base-libs.types
	ext/pango/gsttextoverlay.c
	ext/vorbis/gstvorbisdec.c
	gst/playback/gstplaysink.c
	gst/playback/gstplaysinkconvertbin.c
	sys/ximage/ximagesink.c
	sys/xvimage/xvimagesink.c
 2012-03-08 20:31:34 +00:00
Mark Nauwelaerts
8a3f818dce audiodecoder: add some tag handling convenience help 2012-03-06 16:17:37 +01:00
Mark Nauwelaerts
5a0fff76f3 audiodecoder: add baseclass _CAST macro 2012-03-06 16:17:33 +01:00
Mark Nauwelaerts
d19f5467cc audio: add helper function to convert mask to channel positions 
... as there may be other than raw audio formats using a channel mask,
and there is already one to convert the other way around.
 2012-03-05 13:03:57 +01:00
Mark Nauwelaerts
debbc75272 audioencoder: stop proxying some old-style 0.10 raw audio caps fields 2012-03-05 13:03:57 +01:00
Mark Nauwelaerts
1a2863bf33 audioencoder: store segment event as pending event to forego dropping it 2012-03-05 13:03:57 +01:00
Mark Nauwelaerts
aae64c40a8 audiodecoder: plug caps leak when setting output format 2012-03-05 13:03:57 +01:00
Mark Nauwelaerts
3b0a2a60da audiodecoder: enhance some debug statement 2012-03-05 11:04:20 +01:00
Sebastian Dröge
f7939bb43f Merge branch 'master' into 0.11 
Conflicts:
	NEWS
	RELEASE
	configure.ac
	docs/plugins/gst-plugins-base-plugins.args
	docs/plugins/gst-plugins-base-plugins.hierarchy
	docs/plugins/gst-plugins-base-plugins.interfaces
	docs/plugins/inspect/plugin-adder.xml
	docs/plugins/inspect/plugin-alsa.xml
	docs/plugins/inspect/plugin-app.xml
	docs/plugins/inspect/plugin-audioconvert.xml
	docs/plugins/inspect/plugin-audiorate.xml
	docs/plugins/inspect/plugin-audioresample.xml
	docs/plugins/inspect/plugin-audiotestsrc.xml
	docs/plugins/inspect/plugin-cdparanoia.xml
	docs/plugins/inspect/plugin-encoding.xml
	docs/plugins/inspect/plugin-ffmpegcolorspace.xml
	docs/plugins/inspect/plugin-gdp.xml
	docs/plugins/inspect/plugin-gio.xml
	docs/plugins/inspect/plugin-gnomevfs.xml
	docs/plugins/inspect/plugin-libvisual.xml
	docs/plugins/inspect/plugin-ogg.xml
	docs/plugins/inspect/plugin-pango.xml
	docs/plugins/inspect/plugin-playback.xml
	docs/plugins/inspect/plugin-subparse.xml
	docs/plugins/inspect/plugin-tcp.xml
	docs/plugins/inspect/plugin-theora.xml
	docs/plugins/inspect/plugin-typefindfunctions.xml
	docs/plugins/inspect/plugin-uridecodebin.xml
	docs/plugins/inspect/plugin-videorate.xml
	docs/plugins/inspect/plugin-videoscale.xml
	docs/plugins/inspect/plugin-videotestsrc.xml
	docs/plugins/inspect/plugin-volume.xml
	docs/plugins/inspect/plugin-vorbis.xml
	docs/plugins/inspect/plugin-ximagesink.xml
	docs/plugins/inspect/plugin-xvimagesink.xml
	gst-libs/gst/app/gstappsink.c
	gst-libs/gst/audio/mixer.c
	gst-libs/gst/audio/mixer.h
	gst-libs/gst/tag/gstxmptag.c
	gst-libs/gst/video/colorbalance.c
	gst-libs/gst/video/colorbalance.h
	gst/adder/gstadder.c
	gst/playback/gstplaybasebin.c
	gst/playback/gstplaybin2.c
	gst/playback/gstplaysink.c
	gst/videoscale/gstvideoscale.c
	tests/check/elements/videoscale.c
	tests/examples/seek/seek.c
	tests/examples/v4l/probe.c
	win32/common/_stdint.h
	win32/common/audio-enumtypes.c
	win32/common/config.h
 2012-03-02 10:00:55 +01:00
Wim Taymans
502c12f827 update for metadata API changes 2012-02-29 17:25:10 +01:00
Wim Taymans
a232714065 meta: add return value to transform 2012-02-28 16:18:30 +01:00
Wim Taymans
1c05eeece5 update for metadata tags 2012-02-28 12:10:14 +01:00
Philippe Normand
63ace8872d audio: link against libm 
It is used in gststreamvolume.
 2012-02-27 14:36:25 +00:00
Edward Hervey
59918e841f Suppress deprecation warnings in selected files, for g_value_array_* mostly 2012-02-27 14:28:15 +01:00
Wim Taymans
5a0354b416 audioencoder: don't leak event 2012-02-27 13:08:36 +01:00
Wim Taymans
15eb385412 audioencoder: use default event function 
Implement a default event function so that subclasses can call it without having
to return FALSE (and make it impossible to report errors).
 2012-02-27 12:49:52 +01:00
Wim Taymans
525f330142 update for metadata changes 2012-02-24 10:26:04 +01:00
Wim Taymans
268d52fd33 Merge branch 'master' into 0.11 
Conflicts:
	gst-libs/gst/rtsp/gstrtspconnection.c
	win32/common/libgstaudio.def
 2012-02-17 23:46:17 +01:00
Tim-Philipp Müller
0f6c8a27a7 docs: add new audio base class API to docs and .def file 2012-02-17 15:08:36 +00:00
Wim Taymans
e44dd9db8f Merge branch 'master' into 0.11 
Conflicts:
	gst-libs/gst/audio/gstaudioencoder.c
	gst-libs/gst/pbutils/gstdiscoverer.c
 2012-02-16 14:23:28 +01:00
Mark Nauwelaerts
439884d628 audiodecoder: add some properties to tweak baseclass behaviour 
... so subclass can also rely upon never being bothered with some NULL buffer
it can't do any interesting with, or with any data before it received
any format configuration (and setup properly).
 2012-02-16 12:35:53 +01:00
Mark Nauwelaerts
5b4dc02523 audioencoder: add some properties to tweak baseclass behaviour 
... so subclass can also rely upon never being bothered with less data
than it desires or with some NULL buffer it can't do any interesting with.
 2012-02-16 12:35:51 +01:00
Mark Nauwelaerts
95306e8fef audiodecoder: assert some more that subclass parsed frame has proper len 2012-02-16 12:35:40 +01:00
Wim Taymans
c7d0fb556f audiodecoder: chain up to parent for defaults 
Chain up to the parent instead of using the FALSE return value from
the event function (because it's otherwise impossible to return an error).
 2012-02-15 13:42:19 +01:00
Wim Taymans
b2fbb2e587 audiodecoder: call default event handler 
Call the default event handler for unknown events.
 2012-02-15 13:03:59 +01:00
Wim Taymans
a75e9102c5 GST_FLOW_WRONG_STATE -> GST_FLOW_FLUSHING 2012-02-08 15:17:49 +01:00
Mark Nauwelaerts
97d60612a4 audiodecoder: remove stray obsolete declaration 2012-02-06 22:10:28 +01:00
Mark Nauwelaerts
2bf1a4428e audio: correctly fill in fallback channel positions in stereo case 2012-02-06 22:10:28 +01:00
Wim Taymans
6c08f53416 audiofilter: configure info after calling vmethod 
First call the vmethod and then configure the audioinfo in the baseclass. This
allows subclasses to know about the old format.
 2012-02-06 13:23:26 +01:00
Wim Taymans
fe3e9b90dd audioencoder: don't unref caps parameter 
Fix refcounting on incomming caps to make sure we don't unref it too much.
 2012-02-03 09:51:00 +01:00
Sebastian Dröge
1cb4029d00 audioencoder: gst_pad_get_pad_template_caps() now returns a new reference, don't forget to unref 2012-02-01 16:33:30 +01:00
Sebastian Dröge
5aa6748151 audio{enc,dec}oder: Check if srcpad caps are a subset of the template caps 2012-02-01 16:32:53 +01:00
Sebastian Dröge
0370b0dc12 audioencoder: Add gst_audio_encoder_set_output_format() function for consistency 2012-02-01 16:27:47 +01:00
Sebastian Dröge
dbd43c7dd3 audiodecoder: Rename set_outcaps() to set_output_format() and take a GstAudioInfo as parameter 2012-02-01 16:27:47 +01:00
Wim Taymans
30af2fe7d6 audiosrc: wait on the right cond variable 
This broke with a merge commit
 2012-01-27 18:27:26 +01:00
Wim Taymans
fcdc385aa1 port to new map API 2012-01-25 12:30:53 +01:00
Sebastian Dröge
68c0790817 Merge branch 'master' into 0.11 
Conflicts:
	gst-libs/gst/interfaces/propertyprobe.c
	sys/xvimage/xvimagesink.c
 2012-01-25 11:50:54 +01:00
Wim Taymans
3d42f0f6ed port to new glib thread API 2012-01-19 11:36:17 +01:00
Tim-Philipp Müller
576bbb4fd8 Remove compatibility code cruft for old GLib versions 2012-01-18 17:22:21 +00:00
Mark Nauwelaerts
3e312e6e16 baseaudiosink: commit correct number of samples when not syncing 2012-01-17 21:46:58 +01:00
Mark Nauwelaerts
974c678ec8 audiodecoder: register state change function 2012-01-17 11:53:51 +01:00
Sebastian Dröge
de19cfdd8a audio: More UNPOSITION flag sanity checks 
..and turn the GST_WARNING() into a g_warning(). This is a programming
error and should be fixed.
 2012-01-11 10:49:49 +01:00
Sebastian Dröge
a03f70e3cd audio: Add validity check for the UNPOSITIONED audio flag 
Also reset the flag when parsing caps.
 2012-01-11 10:44:37 +01:00
Sebastian Dröge
05beab5382 audiometa: Improve GstAudioDownmixMeta to be actually usable 
This now has a two-dimensional array of coefficients
as required and also stores the source and destination
channel positions.
 2012-01-10 12:46:05 +01:00
Sebastian Dröge
67c8b0dfbd audio: Don't crash if NULL positions are passed to gst_audio_info_set_format() 2012-01-10 12:02:56 +01:00
Sebastian Dröge
5cb3d75dbf audiobasesink: Fix infinite recursion by chaining up to the correct parent class vfunc 2012-01-09 14:19:54 +01:00
Sebastian Dröge
bb3eb93ee9 audio: Don't check for channel positions in valid order when converting to a channel mask 2012-01-09 08:24:23 +01:00
Edward Hervey
82da418201 audio: Fix size check 
We fail (and return) if the size is *NOT* a multiple of samples.
 2012-01-06 15:14:59 +01:00
Wim Taymans
dd43d0697e audio: expose API to convert channel array to a mask 2012-01-05 13:59:32 +01:00
Sebastian Dröge
9e072ea844 audio: Improve/fix handling of NONE layouts 2012-01-05 10:34:25 +01:00
Sebastian Dröge
8dcea5d498 audio: Add support again for more than 64 channels with NONE layouts 2012-01-05 10:34:25 +01:00
Sebastian Dröge
31c9f7d09a audio: Fix GST_AUDIO_CHANNEL_POSITION_MASK macro 2012-01-05 10:34:25 +01:00
Sebastian Dröge
9d56bf7712 audioencoder: Proxy the channel mask field instead of the old channel-layout field 2012-01-05 10:34:24 +01:00
Sebastian Dröge
8fe5dc53e0 audiocdsrc: Add the layout field to the caps 2012-01-05 10:34:24 +01:00
Sebastian Dröge
810bfec656 audio: Add "layout" field to the raw audio caps 
This can be used to differentiate between interleaved
and non-interleaved audio and whatever comes in the future.
 2012-01-05 10:34:24 +01:00
Sebastian Dröge
e2c6b8ec4d audio: Add function to reorder channel positions from any order to the GStreamer order 2012-01-05 10:34:24 +01:00
Sebastian Dröge
bd40936409 audioringbuffer: Use new function to get a channel reordering map 2012-01-05 10:34:24 +01:00
Sebastian Dröge
9e930a1ade audio: Add documentation for the new functions 2012-01-05 10:34:24 +01:00
Sebastian Dröge
c9c12372a5 audio: Add public functions to check channel positions validity and to get a reorder map 2012-01-05 10:34:24 +01:00
Sebastian Dröge
225238a913 audioringbuffer: Add support for reordering of channels 2012-01-05 10:34:16 +01:00
Sebastian Dröge
c227f5e77e audio: Add new channel positions and simplify channel expression in the caps 
The available channel positions are all channels from SMPTE 2036-2-2008
(in that order) and DTS Coherent Acoustics, which are basically all 28
channels that currently can appear.

The channels are now expressed in the caps as a channel-mask, which
describes which of the channels are present, and an optional
channel-reorder-map, which must only be used after negotiation for
fixated caps.

For negotiation only the channel-mask and the channel count is relevant
and all elements are expected to handle all reorder maps. Elements that
don't can use the new API to reorder an audio buffer from any order to
another order.

This simplifies negotiation a lot while still having as few reorderings
necassary as possible and still allow all kinds of channel layouts.
 2012-01-05 10:27:21 +01:00
Wim Taymans
e9eaf17eae audioencoder: turn assert into a real error 
Post a real error instead of just asserting. Fixes a unit test.
 2012-01-02 15:42:39 +01:00
Tim-Philipp Müller
26e612aeda playback, mixerutils: gst_registry_get_default() -> gst_registry_get() 2012-01-02 14:32:11 +00:00
Wim Taymans
ed6fd4eb2f audio: add flag for unpositioned layout 
Check if thr layout is explicitly unpositioned and set a flag in the
audio info structure.
 2012-01-02 15:01:58 +01:00
Tim-Philipp Müller
c3e6e23b85 audio, rtsp: remove private/protected gtk-doc markup for enums 
This confuses glib-mkenums, and is not really useful anyway.

https://bugzilla.gnome.org/show_bug.cgi?id=666618
 2012-01-02 00:19:57 +00:00
Tim-Philipp Müller
d877ef13f5 docs: make gtk-doc happier 2011-12-30 19:24:09 +00:00
Tim-Philipp Müller
62e5a67376 audiocdsrc: remove some probing-related vfuncs 
GstPropertyProbe was removed, so these aren't actually used
and we probably want something different for the new API.
 2011-12-30 16:26:47 +00:00
Tim-Philipp Müller
6a85353a92 audiocdsrc: update for GstIndex removal 2011-12-30 16:18:39 +00:00
Tim-Philipp Müller
31890ef59b audiocdsrc: make private bits private 2011-12-30 16:12:30 +00:00
Edward Hervey
f562a29284 Merge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	ext/theora/gsttheoraenc.c
	gst-libs/gst/tag/gstexiftag.c
	gst/adder/gstadder.c
	gst/adder/gstadder.h
	gst/playback/gstdecodebin2.c
	gst/playback/gstsubtitleoverlay.c
	tests/check/libs/tag.c
 2011-12-30 13:21:35 +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
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
Wim Taymans
ddc05e0ed1 propertyprobe: remove propertyprobe 
Remove the propertyprobe interface
Improve docs
 2011-12-21 11:58:53 +01:00
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
Vincent Penquerc'h
12be1e6fc5 baseaudiosink: fix late buffer leak 2011-12-13 12:55:45 +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
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
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
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
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
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
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 
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 
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 
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
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
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
Wim Taymans
c42e257751 audio: fix docs 2011-11-11 19:13:52 +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
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 
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 
... 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 
... 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 
... 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
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
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
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
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
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
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
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
Mark Nauwelaerts
35b172004c audioutils: modify _parse, add GType support functions 2011-08-27 14:47:00 +01:00
Mark Nauwelaerts
a4d5e33224 baseaudiodecoder: move properties to private storage and add 
_get/_set
 2011-08-27 14:47:00 +01:00
Mark Nauwelaerts
7939d37936 baseaudiodecoder: rename property 2011-08-27 14:47:00 +01:00
Mark Nauwelaerts
d71e427c49 baseaudiodecoder: replace context helper structure by various 
_get/_set
 2011-08-27 14:47:00 +01:00
Mark Nauwelaerts
a39a66dd4b baseaudioencoder: move properties to private storage and add 
_get/_set
 2011-08-27 14:47:00 +01:00
Mark Nauwelaerts
41a0d6f8f0 baseaudioencoder: rename some properties 2011-08-27 14:47:00 +01:00
Mark Nauwelaerts
6302c9d31d baseaudioencoder: replace context helper structure by various 
_get/_set
 2011-08-27 14:47:00 +01:00
Mark Nauwelaerts
d1ab04f029 baseaudio: rename GstAudioState to GstAudioFormatInfo 2011-08-27 14:47:00 +01:00
Mark Nauwelaerts
ecf57f2b73 baseaudioencoder: TEMP; avoid some imperfect ts jitter ? 
... even when not in perfect mode ?
 2011-08-27 14:47:00 +01:00
Mark Nauwelaerts
5a40343102 baseaudioencoder: debug format fixes 2011-08-27 14:47:00 +01:00
Mark Nauwelaerts
cedbedbbca baseaudiodecoder: debug format fix 2011-08-27 14:47:00 +01:00
Mark Nauwelaerts
8b6109cdbe baseaudiodecoder: fixup documentation 2011-08-27 14:47:00 +01:00
Mark Nauwelaerts
5003868dc7 baseaudiodecoder: fix FLUSH_STOP actions 2011-08-27 14:46:59 +01:00
Mark Nauwelaerts
660aa2e2c0 baseaudiodecoder: preserve upstream seek event seqnum 2011-08-27 14:46:59 +01:00