Commit graph

1125 commits

Author SHA1 Message Date
Vineeth T M
060f0c21f2 audiosink, multisocketsink: Fix error leak during failures
https://bugzilla.gnome.org/show_bug.cgi?id=755143
2015-09-17 11:59:35 +02:00
Tim-Philipp Müller
7dac2e1eb1 audiobasesink: fix misleading error message debug detail
https://bugzilla.gnome.org/show_bug.cgi?id=754260
2015-08-29 10:44:28 +01:00
Carlos Rafael Giani
c95d809a96 audiobasesink: Fix incorrect/missing custom slaving method documentation
https://bugzilla.gnome.org/show_bug.cgi?id=754199
2015-08-28 10:13:44 +03:00
Sebastian Dröge
6cda5d3494 audiodecoder: If there are no tags, don't try to do event handling on a NULL event
Fixes some crashes.
2015-08-18 16:01:28 +03:00
Sebastian Dröge
2de91c32e4 audioencoder: If there are no tags, don't try to do event handling on a NULL event
Fixes some crashes.
2015-08-18 15:58:57 +03:00
Tim-Philipp Müller
4c00709e22 audioencoder: fix tag handling
Merge upstream tags with encoder tags and update whenever
any of those changes.

https://bugzilla.gnome.org/show_bug.cgi?id=679768
2015-08-18 11:46:21 +01:00
Tim-Philipp Müller
135cad5b6e audiodecoder: try harder to avoid sending unnecessary tag updates 2015-08-16 18:28:09 +01:00
Tim-Philipp Müller
5ccc8432e0 audiodecoder: fix tag handling
Before we just merged everything in pretty much random ways
ad-hoc instead of keeping state properly. In 0.10 that was
how it worked, but in 1.x the tag events sent should always
reflect the latest state and replace any previous tags.

So save the upstream (stream) tags, and save the tags set
by the decoder subclass with merge mode, and then update
the merged tags whenever either of those two changes.

This slightly changes the behaviour of gst_audio_decoder_merge_tags()
in case it is called multiple times, since now any call replaces
the previously-set tags. However, it leads to much more predictable
outcomes, and also we are not aware of any subclass which sets this
multiple times and expects all the tags set to be merged.

If more complex tag merging scenarios are required, we'll have
to add a new vfunc for that or the subclass has to intercept
the upstream tags itself and send merged tags itself.

https://bugzilla.gnome.org/show_bug.cgi?id=679768
2015-08-16 16:34:31 +01:00
Thiago Santos
5a0bee3f13 audioencoder: add src and sink query methods
Allows subclasses to do their own handling of GstQuery and still
chain up to the parent class to handle the ones that they don't want
to handle
2015-08-16 08:14:35 -03:00
Thiago Santos
d19f347223 Revert "audiodecoder: expose default query handling function"
Apparently I forgot how gobject works, there is no need to expose
it directly as one can call it from the parent_class pointer

This reverts commit 8a64592481.
2015-08-15 13:44:35 -03:00
Thiago Santos
33b1da7b80 audiodecoder: add option to use default pad accept-caps handling
Add gst_audio_decoder_set_use_default_pad_acceptcaps() to allow
subclasses to make videodecoder use the default pad acceptcaps
handling instead of resorting to the caps query that is, usually,
less efficient and unecessary

API: gst_audio_decoder_set_use_default_pad_acceptcaps
2015-08-15 13:44:34 -03:00
Thiago Santos
8a64592481 audiodecoder: expose default query handling function
Subclasses can use it to select what queries they want to handle
and forward the rest to the default handling function.

API: gst_audio_decoder_sink_query_default

https://bugzilla.gnome.org/show_bug.cgi?id=753623
2015-08-14 13:41:58 -03:00
Hyunjun Ko
9dc463346e video/audio meta: transform_func: return FALSE if not supported or failed
https://bugzilla.gnome.org/show_bug.cgi?id=751778
2015-07-07 16:41:23 +03:00
Sebastian Dröge
b495dfa5e6 {audio,video}info: Add GST_TYPE_{AUDIO,VIDEO}_INFO macros 2015-07-06 12:53:15 +03:00
Sebastian Dröge
56add20dc7 audioencoder: Don't try to get buffers from an empty adapter 2015-07-02 13:16:15 +02:00
Sebastian Dröge
f5eebb27a2 {audio,video}{en,de}oder: Also copy POOL metas and make sure to copy over metas when creating subbuffers
POOL meta just means that this specific instance of the meta is related to a
pool, a copy should be made when reasonable and the flag should just not be
set in the copy.
2015-07-01 10:58:07 +02:00
Sebastian Dröge
844fa28c67 audiodecoder: Add transform_meta() vfunc with default implementation
The default implementation copies all metadata without tags, and metadata
with only the audio tag. Same behaviour as in GstAudioFilter.

https://bugzilla.gnome.org/show_bug.cgi?id=742385
2015-06-30 10:40:10 +02:00
Sebastian Dröge
cc1719130b audioencoder: Add transform_meta() vfunc with default implementation
The default implementation copies all metadata without tags, and metadata
with only the audio tag. Same behaviour as in GstAudioFilter.

https://bugzilla.gnome.org/show_bug.cgi?id=742385
2015-06-30 10:40:10 +02:00
Sebastian Dröge
0acdfeed69 audiofilter: Also copy metas if their API has no tags attached to it
This is the default basetransform behaviour, being more strict than that
is not really useful.
2015-06-29 13:06:33 +02:00
Lyon Wang
829b7298e9 audioringbuffer: Fix alaw/mulaw channel positions
For alaw/mulaw we should also try to initialize the channel positions in the
ringbuffer's audio info. This allow pulsesink to directly use the channel
positions instead of using the default zero-initialized ones, which doesn't
work well.

https://bugzilla.gnome.org/show_bug.cgi?id=751144
2015-06-22 18:29:00 +02:00
Nicolas Dufresne
c101452a4a gi: Use INTROSPECTION_INIT for --add-init-section
This new define was added to common. The new init section fixed
compilation warning found in the init line that was spread across
all files.
2015-06-16 18:04:57 -04:00
Nicolas Dufresne
e8aba7db8b gi: Fix warnings in libgstaudio
* Duplicate section
* Miss-named parameter
* Missing transfer none annotation for meta
2015-06-16 14:34:04 -04:00
Carlos Rafael Giani
c5b75394a9 audiobasesink: added custom clock slaving method
This new clock slaving method allows for installing a callback that is
invoked during playback. Inside this callback, a custom slaving
mechanism can be used (for example, a control loop adjusting a PLL or an
asynchronous resampler). Upon request, it can skew the playout pointer
just like the "skew" method. This is useful if the clocks drifted apart
too much, and a quick reset is necessary.

Signed-off-by: Carlos Rafael Giani <dv@pseudoterminal.org>

https://bugzilla.gnome.org/show_bug.cgi?id=708362
2015-06-09 21:51:05 +10:00
Mathieu Duponchelle
579967278a audiofilter: copy metadata that only has the "audio" tag.
https://bugzilla.gnome.org/show_bug.cgi?id=750406
2015-06-04 19:16:40 +02:00
Sebastian Dröge
0c43005f54 audiodecoder: Also negotiate with downstream if needed before handling a GAP event 2015-05-08 15:07:56 +02:00
Sebastian Dröge
9e480d11a2 audio: Change the remaining "samples" in the ::delay() vfunc docs to "frames"
https://bugzilla.gnome.org/show_bug.cgi?id=748289
2015-04-27 10:08:17 +02:00
Sebastian Dröge
531b9ba951 audio: The delay vfunc returns the number of frames, not samples
https://bugzilla.gnome.org/show_bug.cgi?id=748289
2015-04-26 21:08:14 +02:00
Tim-Philipp Müller
c680e324bc Remove obsolete Android build cruft
This is not needed any longer.
2015-04-26 18:42:34 +01:00
Wonchul Lee
5dffb8a311 audiodecoder: Add sink and src query virtual method
API: GstAudioDecoderClass::src_query()
API: GstAudioDecoderClass::sink_query()

https://bugzilla.gnome.org/show_bug.cgi?id=747293
2015-04-23 19:29:20 +01:00
Vincent Penquerc'h
5cb40d7320 audiobasesink: fix ring buffer leak on open failure 2015-04-09 13:00:58 +01:00
Vincent Penquerc'h
4e6b917ba3 audiobasesrc: fix ring buffer leak on open failure 2015-04-09 13:00:57 +01:00
Sebastian Dröge
a21795260f audiodecoder: Don't post error messages while holding the stream lock 2015-04-08 20:49:39 -07:00
Sebastian Dröge
9196c3dcca audiodecoder: Don't get and parse the current srcpad caps
We only get here if we don't have any srcpad caps, and we're going
to override the GstAudioInfo a few lines below anyway without ever
using it if for whatever reason we get caps here.
2015-04-08 20:49:39 -07:00
Sebastian Dröge
0c72d0acdf {audio,video}decoder: Forward SEGMENT_DONE events immediately and drain decoders
Otherwise we're going to wait with draining until the next data comes, which
is a bit suboptimal and might take a long time... or maybe never happens.
2015-04-06 19:20:51 -07:00
Vincent Penquerc'h
2954813b86 audio,video: use gst_segment_is_equal instead of memcmp
memcmp will blindly compare the reserved fields, as well as any
padding the compiler may choose to sprinkle in GstSegment.

Fixes valgrind complaints in unit tests, as well as some found via
https://bugzilla.gnome.org/show_bug.cgi?id=738216
2015-04-03 12:09:41 +01:00
Edward Hervey
3eb35c77cc introspection: Don't use g-ir-scanner cache at compile time
It pollutes user directories and we don't need to cache it

https://bugzilla.gnome.org/show_bug.cgi?id=747095
2015-03-31 11:21:43 +02:00
Arun Raghavan
592fc9cdba audioringbuffer: Log with the ringbuffer object where possible 2015-03-13 23:24:58 +05:30
Mark Nauwelaerts
eeeb2eab82 audiodecoder: only return EOS upon clipping if applicable
See also https://bugzilla.gnome.org/show_bug.cgi?id=709224
2015-03-07 20:10:31 +01:00
Arun Raghavan
557c2c9be1 audiobasesink: Reset audio clock if necessary
When the ringbuffer is deactivated and then acquired, if the audio clock
provided by the sink gets reset to zero, we need to add an offset to the
clock to make sure that subsequent samples are written out at the right
times. While we need to leave this to derived classes to take care of
when they provide their own clock (since that clock may or may not be
reset to zero), we can do this ourselves if we know the provided clock
is our own (which does reset to zero on a re-acquire).
2015-03-03 23:26:54 +05:30
Jan Schmidt
b3053925ac audiodecoder: Don't send pending events before decode
Make sure to update the output segment to track the segment
we're decoding in, but don't actually push it downstream until
after buffers are decoded.

https://bugzilla.gnome.org/show_bug.cgi?id=744806
2015-02-24 01:36:44 +11:00
Mark Nauwelaerts
c321b6bd81 Revert "audiodecoder: drain current segment upon new one to ensure correct flow return"
This reverts commit 696b8cdc40.

See https://bugzilla.gnome.org/show_bug.cgi?id=734617
2015-02-22 16:58:33 +01:00
Mark Nauwelaerts
696b8cdc40 audiodecoder: drain current segment upon new one to ensure correct flow return
See also https://bugzilla.gnome.org/show_bug.cgi?id=709224
2015-02-22 13:23:44 +01:00
Thiago Santos
7e39a51a50 audio: video: fix a few GI annotations
transfer-full -> transfer full
@Since -> Since
2015-02-19 15:51:42 -03:00
Sebastian Dröge
8547594727 Improve and fix LATENCY query handling
This now follows the design docs everywhere, especially the maximum latency
handling.

https://bugzilla.gnome.org/show_bug.cgi?id=744106
2015-02-11 17:53:49 +02:00
Jan Schmidt
4f961e6d95 audiodecoder: Where possible, skip decode for GST_SEGMENT_FLAG_TRICKMODE_NO_AUDIO
If we have timestamps on input buffers and are in trickmode no-audio
mode, then don't pass anything to the subclass for decode and simply
send gap events downstream

Only for forward playback for now - reverse requires accumulating
GAP events and pushing out in reverse order.

https://bugzilla.gnome.org/show_bug.cgi?id=735666
2015-02-06 04:09:37 +11:00
Jan Schmidt
ca231ce321 audiobasesink: Re-work GAP buffer and trick-mode handling
In trickmode no-audio mode, or when receiving a GAP buffer,
discard the contents and render as a GAP event instead.

Make sure when rendering a gap event that the ring buffer will
restart on PAUSED->PLAYING by setting the eos_rendering flag.

This mostly reverts commit 8557ee and replaces it. The problem
with the previous approach is that it hangs in wait_preroll()
on a PLAYING-PAUSED transition because it doesn't commit state
properly.

https://bugzilla.gnome.org/show_bug.cgi?id=735666
2015-02-06 04:09:37 +11:00
Jan Schmidt
c35e3e7c7d audiodecoder: Remove pointless else{} around some code 2015-02-06 04:02:48 +11:00
Jan Schmidt
7c0f885ad2 audiodecoder: Fix reverse playback when there's only one gather set.
The decoder can fail to drain on EOS if there was only one gather
set, because it will never have sent the segment event downstream
and set the output segment, and fail to detect that the rate < 0.0

Make sure to send pending events before sending all the gather data
for decode.
2015-02-06 04:02:48 +11:00
Sebastian Dröge
823cb40642 audio{enc,dec}oder: Always directly post latency messages on the bus when the subclass sets the latency
Instead of doing it only in setcaps for the encoder, and never at all for the
decoder.
2015-02-03 12:15:25 +01:00
Sebastian Dröge
f2a762a3a0 audio{enc,dec}oder: Handle max_latency == GST_CLOCK_TIME_NONE
And initialize the latencies with 0 and NONE.
2015-02-03 12:12:18 +01:00