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3409 commits

Author SHA1 Message Date
Nicolas Dufresne
7107e97273 videoencoder: Remove done ToDo 
https://bugzilla.gnome.org/show_bug.cgi?id=675761
 2012-12-31 19:09:01 +00:00
Nicolas Dufresne
8a233a215d videoencoder: Documentation fix 
https://bugzilla.gnome.org/show_bug.cgi?id=675761
 2012-12-31 19:03:29 +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
Tim-Philipp Müller
a3c6d0da91 encoding-profile: add special-casing for asf/wmv/wma file extensions 
https://bugzilla.gnome.org/show_bug.cgi?id=636753
 2012-12-23 15:51:51 +00:00
Tim-Philipp Müller
42f971c5eb encoding-profile: add gst_encoding_profile_get_file_extension() 
API: gst_encoding_profile_get_file_extension()

https://bugzilla.gnome.org/show_bug.cgi?id=636753
 2012-12-23 15:26:59 +00:00
Tim-Philipp Müller
df186d5240 video: fix A420 size calculation 2012-12-22 21:04:11 +00:00
Wim Taymans
ca456ec6f9 riff: add channel masks for all formats 
Add the channel masks for all the extensible formats
Pass the number of channels instead of reading them from caps.
 2012-12-21 14:03:32 +01:00
Pete Beardmore
d2c68e602d riff: add waveformatextension ac3 support 
fixes #690591
 2012-12-21 13:28:41 +01:00
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
Thijs Vermeir
675562d362 video: use appropriate printf format for gsize 2012-12-18 15:31:52 +01:00
Thijs Vermeir
2887485358 rtp: fix compiler warning 
comparison is always true due to limited range of data type
 2012-12-18 15:27:48 +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
Thibault Saunier
e79f0e801e encodebing: Use the preset_name as the factory name and preset as the name of the preset 
The naming is not perfect, but at least we can keep the exact same behaviour as
before.
 2012-12-17 10:12:11 -03:00
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
Tim-Philipp Müller
3f583351d4 pbutils: add some more flags and file extensions to internal media type descriptions table 
For later use.

https://bugzilla.gnome.org/show_bug.cgi?id=636753
https://bugzilla.gnome.org/show_bug.cgi?id=549111
 2012-12-15 14:08:26 +00:00
Wim Taymans
65c5ecd270 rtspconnection: add limit to queued messages 
Add a limit to the amount of queued bytes or messages we allow on the watch.

API: GstRTSPConnection::gst_rtsp_watch_set_send_backlog()
API: GstRTSPConnection::gst_rtsp_watch_get_send_backlog()
 2012-12-14 11:36:58 +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
e05abf0ef1 docs: fix up some more GstXOverlay -> GstVideoOverlay 
https://bugzilla.gnome.org/show_bug.cgi?id=689740
 2012-12-10 13:40:26 +00:00
Sebastian Dröge
0bb5c6c012 videodecoder: Only keep track of timestamps if the subclass is parsing data 
Otherwise we just pass through the timestamps directly and don't
need to waste additional memory for them.

Fixes bug #689814.
 2012-12-10 11:51:02 +00:00
Sebastian Rasmussen
d4b6f3c1a0 rtspmessage: Add several missing g-i annotations 
https://bugzilla.gnome.org/show_bug.cgi?id=689873
 2012-12-10 10:58:12 +01:00
Thibault Saunier
7358cba017 encodebin: Make use of the new preset_name when setting a preset 
The behaviour is sensibly changed here. Instead of purely falling when a
preset is set on the #GstEncodingProfile, we now make sure that the
element that is plugged corresponds to the one specified as preset. Then,
if we have a preset_name, we use it, if it fails, we fail (we might rather
just keep working even without setting the element properties?)

 + Add tests that it behave correctly
 2012-12-05 17:48:38 -03:00
Thibault Saunier
6a7f688939 encoding-profile: Let the user decide what preset name to use 
It was possible to decide only what #GstElement implementing #GstPreset
to use during the encoding, we can now let the user select a specific preset previously
saved using #gst_preset_save_preset specifying the name chosen when it was saved
in the gst_encoding_profile_set_preset_name.

Actually loading a preset with %NULL as a name would have always failed, so
in the current state of the API that feature is unusable

API:
  gst_encoding_profile_set_preset_name
  gst_encoding_profile_get_preset_name
 2012-12-05 17:36:21 -03:00
Thiago Santos
26d72a73f5 pbutils: encoding-profile: fix _new function introspection docs 
Makes the parameter accept NULL as input for GI bindings
 2012-12-04 13:19:26 -03:00
Tim-Philipp Müller
fbff6c6fb1 audioencoder: add some more debug info and remove obsolete comment 2012-12-02 12:33:43 +00:00
Wim Taymans
b511f938d4 rtsp: add method to parse options list 2012-11-27 11:15:34 +01:00
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
Tim-Philipp Müller
020eb24dcf app: fix g-i annotation for gst_app_src_push_buffer() 
It takes ownership of the buffer.
 2012-11-21 21:53:13 +00:00
Wim Taymans
ce904ec551 rtsprange: add string conversion for new formats 2012-11-21 16:25:24 +01:00
Wim Taymans
fdf904db32 rtsprange: add method to convert ranges to GstClockTime 
Add a method to convert the values of GstRTSPRange to GstClockTime.
Add unit tests for the conversions.

API: gst_rtsp_range_get_times()
 2012-11-21 15:35:46 +01:00
Wim Taymans
f1669d7d9c range: don't overwrite unit field 2012-11-21 15:29:05 +01:00
Wim Taymans
0bf50cd3d8 range: add g_return_if check 2012-11-21 15:29:05 +01:00
Sebastian Dröge
7af386fdaf libs: Fix last commit by using correct include paths and only include existing headers 2012-11-21 11:12:57 +01:00
Evan Nemerson
4d77fba46c libs: Add missing single include headers and use them in GIRs 2012-11-21 11:01:24 +01:00
Wim Taymans
a87cd40f49 rtsprange: improve docs 2012-11-21 10:25:51 +01:00
Sebastian Dröge
15ee41dfc9 discoverer: Add support for getting the stream-id 
https://bugzilla.gnome.org/show_bug.cgi?id=654830
 2012-11-20 14:57:42 +01:00
Sebastian Dröge
e223e313b6 discoverer: Use switch/case instead of lots of ifs for the event handling 2012-11-20 14:37:51 +01:00
Sebastian Dröge
1990c45b60 videodecoder: Return the proportion directly 2012-11-20 12:21:08 +01:00
Sebastian Dröge
6228872df7 videodecoder: Rename from get_qos_info() to get_qos_proportion() 
And only return the proportion. The earliest time already can be
retrieved from get_max_decode_time() and by renaming we allow this
to be more extensible in the future.
 2012-11-20 12:08:26 +01:00
Wim Taymans
b785c66098 rtsp: avoid ABI break 
Move new fields into structures appended at the end of the GstRTSPRange
to avoid ABI break.
 2012-11-20 11:13:01 +01:00
Alessandro Decina
9042efd458 pbutils: fix transfer annotation for gst_encoding_profile_set_restriction 2012-11-20 07:17:53 +01:00
Andoni Morales Alastruey
5f55ea1ef3 videodecoder: add getter for QoS proportion and earliest_time 
Add a getter for the QoS proportion and earliest_time to help
subclasses do better estimations based on the proportion.

API: gst_video_decoder_get_qos_info()

https://bugzilla.gnome.org/show_bug.cgi?id=687991
 2012-11-19 23:57:43 +00:00
Wim Taymans
41d36b2584 rtsp: fix format string 2012-11-19 17:08:38 +01:00
Wim Taymans
fe4b415f98 rtsp: parse UTC ranges 2012-11-19 16:59:48 +01:00
Wim Taymans
b113f9697a rtsp: parse SMPTE ranges 2012-11-19 16:15:46 +01:00
Wim Taymans
02a5940a45 range: handle parse errors better 2012-11-19 16:13:56 +01:00
Wim Taymans
84b1ee4987 rtsp: detect npt time parse errors 2012-11-19 16:04:01 +01:00
Wim Taymans
25580430b0 range: a single - is not allowed 2012-11-19 13:56:53 +01:00
Wim Taymans
db7ea32f35 range: handle ranges starting with - 
An RTSP range that starts with a - means that the first value of the range is
the end of the stream.
 2012-11-19 13:56:53 +01:00