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811 commits

Author SHA1 Message Date
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
Mark Nauwelaerts
d1f5c34fe7 baseaudioencoder: use buffer running time for granule calculation 2011-08-27 14:46:59 +01:00
Mark Nauwelaerts
6c04035eec baseaudiodecoder: minor fix in ts resync 2011-08-27 14:46:59 +01:00
Mark Nauwelaerts
d46006b198 baseaudiodecoder: improve glitch resilience 
Provide a replacement for GST_ELEMENT_ERROR to avoid aborting at the first
atom out of place, while on the other hand not failing indefinitely.
 2011-08-27 14:46:59 +01:00
Mark Nauwelaerts
79b41f59f6 baseaudiodecoder: add limited legacy seeking support 2011-08-27 14:46:59 +01:00
Mark Nauwelaerts
0c33df6540 baseaudiodecoder: cater for audio-codec tag 2011-08-27 14:46:59 +01:00
Mark Nauwelaerts
1dbbe7c89d baseaudiodecoder: initial version 2011-08-27 14:46:59 +01:00
Mark Nauwelaerts
87409f2587 baseaudioencoder: misc fixes 2011-08-27 14:46:59 +01:00
Mark Nauwelaerts
8c61685554 baseaudio: add audioutils for caps and query handling helper utils 2011-08-27 14:46:59 +01:00
Mark Nauwelaerts
cb04eaaa8f baseaudioencoder: mark unstable API 2011-08-27 14:46:59 +01:00
Mark Nauwelaerts
b47c08ba17 baseaudioencoder: fix clearing context 2011-08-27 14:46:59 +01:00
Mark Nauwelaerts
e3cae1619c baseaudioencoder: simplify latency variable handling 2011-08-27 14:46:59 +01:00
Mark Nauwelaerts
9ce2edc918 baseaudioencoder: minor fixes and code simplifications 
Also modify and elaborate a bit on pre_push (though currently unused to no harm).
 2011-08-27 14:46:59 +01:00
Mark Nauwelaerts
d0e9fbf3db baseaudioencoder: additional documentation on granule semantics and 
configuration
 2011-08-27 14:46:59 +01:00
Mark Nauwelaerts
9f7849eac9 baseaudioencoder: elaborate property names 2011-08-27 14:46:58 +01:00
Mark Nauwelaerts
bf61f04577 baseaudioencoder: rename state field xint to is_int 2011-08-27 14:46:58 +01:00
Mark Nauwelaerts
3d2f496b3a baseaudioencoder: gtk-doc syntax fixes 2011-08-27 14:46:58 +01:00
Mark Nauwelaerts
51acb02342 baseaudioencoder: minor fix and cleanup 2011-08-27 14:46:58 +01:00
Mark Nauwelaerts
90d99f23c6 baseaudiocodec: ... and also rename to baseaudiodecoder 2011-08-27 14:46:58 +01:00
Mark Nauwelaerts
dfd7616f60 gst-libs/gst/audio: Remove baseaudiodecoder 
Adds little beyond baseaudiocodec (seeking, bit of query), and what it adds
is mainly out-of-scope (e.g. decoder seeking, should be done by upstream
demuxer/parser) and/or based on non-prime example (mad).
 2011-08-27 14:46:58 +01:00
Iago Toral
492ab47fd2 baseaudiodecoder: Return TRUE if we run into special conversion cases. 2011-08-27 14:46:50 +01:00
Iago Toral
2ed1331f43 audio: initial version of GstBaseAudioCodec 
Moved most of the code to GstBaseAudioCodec, GstBaseAudioDecode is
now really small, maybe we do not really need it (or its encoder
counterpart). Added more API for subclasses and documentation.
 2011-08-27 14:45:47 +01:00
Iago Toral
9740eb35b8 Added src_queries to decoder class. Added handle_discont to decoder 
class. Reworked reset. Various other minor fixes.
 2011-08-27 14:45:47 +01:00
Iago Toral
d05c805b16 Added a draft implementation of gstbaseaudiodecoder 2011-08-27 14:45:47 +01:00
Mark Nauwelaerts
fc6b421227 Added audio directory for audio codec base classes 2011-08-27 14:45:47 +01:00
Mark Nauwelaerts
ef92c7438d audioencoders: add streamheader helper utility 2011-08-27 14:45:47 +01:00
Mark Nauwelaerts
80241fde8d audioencoders: baseaudioencoder and ported encoders 2011-08-27 14:45:47 +01:00
Wim Taymans
6854f2bbf1 multichannel: add some more channels 2011-08-24 18:39:47 +02:00
Wim Taymans
24ea19935f audio/video: add format of the pack functions 
Replace the unpack_size with an unpack_format, which is more descriptive of the
kind of data the unpack function will create.
 2011-08-24 16:40:43 +02:00
Wim Taymans
0a1874461a audio: rename UNPOSITIONED to DEFAULT_POSITIONS 
Rename the UNPOSITIONED flag to the DEFAULT_POSITIONS flag because that is
really what the resulting GstAudioInfo will contain as the chanel mappings.
 2011-08-24 14:13:33 +02:00
Wim Taymans
c6758ecfa9 audio: move function to convert 2011-08-22 16:11:27 +02:00
Wim Taymans
3fab57b5cf Merge branch 'master' into 0.11 
Conflicts:
	gst-libs/gst/interfaces/videooverlay.c
	gst-libs/gst/rtp/gstrtpbuffer.c
	po/af.po
	po/az.po
	po/bg.po
	po/ca.po
	po/cs.po
	po/da.po
	po/de.po
	po/el.po
	po/en_GB.po
	po/es.po
	po/eu.po
	po/fi.po
	po/fr.po
	po/gl.po
	po/hu.po
	po/id.po
	po/it.po
	po/ja.po
	po/lt.po
	po/lv.po
	po/nb.po
	po/nl.po
	po/or.po
	po/pl.po
	po/pt_BR.po
	po/ro.po
	po/ru.po
	po/sk.po
	po/sl.po
	po/sq.po
	po/sr.po
	po/sv.po
	po/tr.po
	po/uk.po
	po/vi.po
	po/zh_CN.po
 2011-08-22 13:06:27 +02:00
Stefan Kost
01bbdd6bdf docs: handle warnings emitted by gtk-doc 
This is useful and in most cases someone had put arbitrary markup into the docs,
misspelled xref'ed symbols, forgot to add stuff to the docs etc..
 2011-08-20 19:16:42 +02:00
Wim Taymans
0213407fbc audio: rename INT -> INTEGER 
Spell INTEGER fully instead of using the int abreviation.
Remove some old functions.
 2011-08-20 10:49:17 +02:00
Wim Taymans
7db6fa37b4 audio: add function to build audio format 2011-08-19 16:00:33 +02:00
Wim Taymans
17dd31b0f4 audio: add more macros 2011-08-19 14:03:23 +02:00
Sebastian Dröge
85a3e7c98c audiofilter: Pass a const pointer to the audio format info to ::setup() 
It is not meant to be changed by the subclass.
 2011-08-19 10:06:39 +02:00
Wim Taymans
dae848818d audio: rework audio caps. 
Rework the audio caps similar to the video caps. Remove
width/depth/endianness/signed fields and replace with a simple string
format and media type audio/x-raw.
Create a GstAudioInfo and some helper methods to parse caps.
Remove duplicate code from the ringbuffer and replace with audio info.
Use AudioInfo in the base audio filter class.
Port elements to new API.
 2011-08-18 19:15:03 +02:00
Wim Taymans
d1a83d7a41 baseaudiosrc: chain up to parent in fixate 2011-08-17 17:24:35 +02:00
Wim Taymans
33467d9629 Merge branch 'master' into 0.11 
Conflicts:
	configure.ac
	ext/pango/gsttextoverlay.c
	ext/theora/gsttheoradec.c
	gst/adder/gstadder.c
	gst/adder/gstadder.h
	gst/audioresample/gstaudioresample.c
	gst/encoding/gstencodebin.c
	gst/playback/gstdecodebin.c
	gst/playback/gstdecodebin2.c
	tests/check/elements/decodebin2.c
	tests/check/elements/playbin-compressed.c
	win32/common/libgsttag.def
 2011-08-16 18:01:14 +02:00
Wim Taymans
d6740006d4 audio: remove deprecated methods 2011-08-16 16:59:15 +02:00
Josep Torra
5629ed74b3 Fix debug statements 
Fixes build on MacOSX

Signed-off-by: Edward Hervey <edward.hervey@collabora.co.uk>
 2011-08-10 11:15:41 +02:00
Wim Taymans
86a10fbb9f baseaudiosrc: call parent alloc function 
Call the parent alloc function to allocate buffers.
 2011-08-04 18:08:49 +02:00
Stefan Sauer
264d91a502 baseaudiosink: fix latency calculation for live elements 
Max_latency was computed on already adjusted min_latency. Introduce a new
variable for clarity. Spotted by Blaise Gassend.
Fixes #644284
 2011-07-28 14:31:47 +02:00
Mark Nauwelaerts
68231a645a baseaudiosink: fix max latency calculation 
... to allow infinite max, as also claimed by comment.
 2011-07-28 12:05:06 +02:00
Mark Nauwelaerts
5d0f279fea baseaudiosink: drop samples that are too late 
... rather than having all of them rendered at 0 or subsequently aligned,
likely inevitably leading to repeated resyncing.
 2011-07-28 11:47:52 +02:00
Wim Taymans
a3971d2afe baseaudiosink: chain up to parent_class correctly 2011-07-26 12:42:22 +02:00
Wim Taymans
8aea5d34bd baseaudiosink: use new basesink query vmethod 2011-07-26 12:37:04 +02:00
Tim-Philipp Müller
4bf26ba5d2 Add -DGST_USE_UNSTABLE_API to the compiler flags to avoid warnings 2011-07-05 10:07:08 +01:00
Wim Taymans
a58805216a audio: clean up headers 2011-06-21 18:17:59 +02:00
Wim Taymans
2e837743c3 audio: clean up audiosink headers 2011-06-21 18:13:48 +02:00
Wim Taymans
d9e1e23094 audio: clean up ringbuffer header 2011-06-21 18:08:12 +02:00
Wim Taymans
f9967e4aac Merge branch 'master' into 0.11 
Conflicts:
	gst-libs/gst/video/video.h
	gst/playback/gstplaysinkaudioconvert.c
	gst/playback/gstplaysinkvideoconvert.c
	tests/check/libs/rtp.c
 2011-06-02 12:18:13 +02:00
Stefan Kost
940291dd38 audio: move testchannels example to 'tests/examples' dir 
Also fix it up a little to not include 'c' file but link to the libs instead.
 2011-05-27 15:09:25 +03:00
Wim Taymans
e614c6bd81 feature: use object name instaed of feature name 2011-05-24 18:21:06 +02:00
Wim Taymans
010add200a scheduling: port to new scheduling query 2011-05-24 17:37:45 +02:00
Wim Taymans
a87c021237 Merge branch 'master' into 0.11 
Conflicts:
	gst-libs/gst/video/convertframe.c
 2011-05-24 09:47:15 +02:00
Stefan Kost
6bee2cb4ee docs: add missing documentation for various pieces 2011-05-23 23:56:09 +03:00
Thijs Vermeir
dad50ad1fe baseaudiosink: recalibrate clock on setcaps 
Because the spec for the ringbuffer can change when changing
the caps, we must recalibrate the clock.

https://bugzilla.gnome.org/show_bug.cgi?id=610443
 2011-05-23 17:02:03 +02:00
Stefan Kost
089fdb7792 docs: fixup audio-library docs 2011-05-23 15:08:24 +03:00
Stefan Kost
d6ea8d5cb3 docs: fix docs for new api 
Some parameters where wrong, first line missed the ':' and return docs where
broken.
 2011-05-23 14:56:17 +03:00
Sebastian Dröge
8a0bdbf2bc audiofilter: gst_pad_template_new() does not take ownership of the caps anymore 
There's no need to copy the caps before passing them to that function.
 2011-05-17 12:31:18 +02:00
Sebastian Dröge
318ed07598 Revert "-base_port to new query API" 
This reverts commit c9f4e0676b.
 2011-05-17 11:25:31 +02:00
Sebastian Dröge
d0362c2b87 Merge branch 'master' into 0.11 
Conflicts:
	configure.ac
	ext/alsa/gstalsasrc.c
	gst-libs/gst/audio/gstbaseaudiosink.c
	gst-libs/gst/tag/gstxmptag.c
	gst/playback/gstsubtitleoverlay.c
	gst/videorate/gstvideorate.c
	sys/xvimage/xvimagesink.c
 2011-05-16 17:06:22 +02:00
Wim Taymans
94dfe80f71 -base: port to new SEGMENT API 2011-05-16 13:48:11 +02:00
Arun Raghavan
623e8781ab baseaudiosink: Use g_str_equal() instead of strncmp() 
The strncmp is unnecessary anyway since one of the strings is a const
string.
 2011-05-14 18:53:12 +05:30
Arun Raghavan
824e643ec9 baseaudiosink: Fix trivial indentation problems 2011-05-14 18:53:12 +05:30
Arun Raghavan
8ff93a6a3d audio: Add an IEC 61937 payloading library 
This can be used by sinks to take compressed formats, correctly payload
these in IEC 61937 frames and feed these to sinks that support
passthrough output over IEC 60958 (S/PDIF) or, in the case of MP3, over
Bluetooth.

Initial implementation includes AC3, E-AC3, MPEG-1, MPEG-2 (non-AAC),
and DTS (type-I/II/II) payloading. More formats can be added as needed.

API: gst_audio_iec61937_frame_size()
API: gst_audio_iec61937_payload()

https://bugzilla.gnome.org/show_bug.cgi?id=642730
 2011-05-14 18:53:12 +05:30
Arun Raghavan
643e5f586c baseaudiosink: Allow subclasses to provide payloaders 
This allows subclasses to provide a "payload" function to prepare
buffers for consumption. The immediate use for this is for sinks that
can handle compressed formats - parsers are directly connected to the
sink, and for formats such as AC3, DTS, and MPEG, IEC 61937 patyloading
might be used.

API: GstBaseAudioSinkClass:payload()

https://bugzilla.gnome.org/show_bug.cgi?id=642730
 2011-05-14 18:23:18 +05:30
Arun Raghavan
9615081f9c ringbuffer: Add support for E-AC3 
Adds support for pushing E-AC3 buffers and doing bytes-to-ms conversion
correctly. The assumption (as with other formats) is that something like
IEC 61937 payloading will be used. Correspondingly the ringbuffer spec
is populated so that the data rate is 4x normal AC3.

https://bugzilla.gnome.org/show_bug.cgi?id=642730
 2011-05-14 18:21:23 +05:30
Arun Raghavan
193fbf93a9 ringbuffer: Add support for MPEG audio buffers 2011-05-14 18:21:16 +05:30
Arun Raghavan
1a1f2cc50a ringbuffer: Add AAC format types 
These are meant to be used for buffers containing AAC data. Nothing uses
this yet, but for now it serves to distinguish from GST_BUFTYPE_MPEG
which represents non-AAC MPEG audio.

API: GST_BUFTYPE_MPEG2_AAC
API: GST_BUFTYPE_MPEG4_AAC
 2011-05-14 18:20:37 +05:30
Arun Raghavan
33ef9ab054 ringbuffer: Add support for DTS buffers 2011-05-14 16:53:33 +05:30
Wim Taymans
c9f4e0676b -base_port to new query API 2011-05-10 18:39:07 +02:00
Wim Taymans
816f4e791d segment: fix for new core API 
Fix for gst_*_segment_full rename.
 2011-05-09 18:16:46 +02:00
Wim Taymans
ec57868488 -base: don't use buffer caps 
Port to newest 0.11 core API, remove GST_PAD_CAPS and GST_BUFFER_CAPS.
 2011-05-09 13:05:12 +02:00
Sebastian Dröge
68a3828adb audiofilter: GstElement takes ownership of pad templates and it should be called from class_init now, not base_init 2011-04-19 14:31:20 +02:00
Sebastian Dröge
f50b3af5d7 audio: Use G_DEFINE_TYPE instead of GST_BOILERPLATE 2011-04-19 10:52:00 +02:00
Sebastian Dröge
0759ce8533 Merge branch 'master' into 0.11 2011-04-18 13:23:32 +02:00
Håvard Graff
d9f1b3736e ringbuffer: make sure to not start if the may_start flag is FALSE 
Fixes #635784
 2011-04-18 11:40:06 +02:00
Sebastian Dröge
c8792778f8 Merge branch 'master' into 0.11 2011-04-16 16:06:26 +02:00
Tim-Philipp Müller
1d05e81435 libs: gobject-introspection scanner doesn't need to scan or update plugin info 
Make sure the scanner doesn't load or introspect or check any plugins,
(especially not outside the build directory).
 2011-04-16 11:01:53 +01:00
Wim Taymans
6e160bed3d Merge branch 'master' into 0.11 
Conflicts:
	android/alsa.mk
	android/app.mk
	android/app_plugin.mk
	android/audio.mk
	android/audioconvert.mk
	android/decodebin.mk
	android/decodebin2.mk
	android/gdp.mk
	android/interfaces.mk
	android/netbuffer.mk
	android/pbutils.mk
	android/playbin.mk
	android/queue2.mk
	android/riff.mk
	android/rtp.mk
	android/rtsp.mk
	android/sdp.mk
	android/tag.mk
	android/tcp.mk
	android/typefindfunctions.mk
	android/video.mk
 2011-04-11 11:37:51 +02:00
Alessandro Decina
030f639a8e android: make it ready for androgenizer 
Remove the android/ top dir
Fixe the Makefile.am to be androgenized

To build gstreamer for android we are now using androgenizer which generates the
needed Android.mk files.
Androgenizer can be found here:
http://git.collabora.co.uk/?p=user/derek/androgenizer.git
 2011-04-11 07:23:21 +02:00
Wim Taymans
da1c863711 Merge branch 'master' into 0.11 
Conflicts:
	gst-libs/gst/tag/gstvorbistag.c
 2011-04-04 11:31:33 +02:00
Stian Johansen
0f8edca902 baseaudiosrc: Add src object lock around call to ringbuffer parse caps. 
A race was observed between query() and setcaps() where the latter would
change the ringbuffer spec while the former was performing operations
based this data.
 2011-04-04 09:35:58 +02:00
Havard Graff
63cfa2a50d baseaudiosrc: protect against ringbuffer disappearing while in a query 
Observed a case where the src went to null-state during the query,
hence the spec pointer was no longer valid, and
gst_util_unit64_scale_int crashed (assertion `denom > 0´failed)

Add locking to make sure the ringbuffer can't disappear.
 2011-04-04 09:33:33 +02:00
Havard Graff
588ac0ae6f baseaudiosink: don't allow aligning behind the read-segment 
Given a large enough drift-tolerance, one could end up in a situation
where one would keep aligning the written buffers behind the current
read-segment position. The result for the reader would be complete
silence, possible preceded by very choppy audio.

By checking the available headroom, one can determine if there is
room to do alignment, or if one should resort to a resync instead to get
the pointers back on track.

Also refactor the alignment-logic out of the render function for cleaner
code.
 2011-04-04 09:31:26 +02:00