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Author SHA1 Message Date
Tim-Philipp Müller
6ff3dfe946 matroskamux: don't forward tag events downstream 
Don't forward stream-specific tag events downstream (esp. not
before any newsegment event).x
 2010-10-14 12:48:00 +01:00
Mark Nauwelaerts
c7a8d672a7 qtdemux: handle another mp4v variation 
... including the glbl atom containing codec-data.
 2010-10-13 17:26:33 +02:00
Stefan Kost
d8167e3071 various (gst): add a missing G_PARAM_STATIC_STRINGS flags 2010-10-13 18:00:28 +03:00
Stefan Kost
77b656eec1 various (sys): add a missing G_PARAM_STATIC_STRINGS flags 2010-10-13 17:39:36 +03:00
Stefan Kost
45f6707784 various (ext): add a missing G_PARAM_STATIC_STRINGS flags 2010-10-13 17:39:36 +03:00
Stefan Kost
2975307f86 various: wrap property registration and add a single fixme for long desc. 2010-10-13 17:39:21 +03:00
Wim Taymans
9f8b56b974 h264depay: always mark the codec_data as keyframe 
We need to mark the codec_data as a keyframe or else downstream decoders might
decide to skip it, waiting for a keyframe.

Fixes #631996
 2010-10-13 11:48:49 +02:00
Zaheer Abbas Merali
6f0030f701 matroskamux: make buffer offsets a byte count rather than a buffer count 2010-10-13 07:17:24 +01:00
Tim-Philipp Müller
d65eb2b91a ext, gst: canonicalise property names where this wasn't the case 
ie. "foo_bar" -> "foo-bar"
 2010-10-12 16:04:21 +01:00
Thijs Vermeir
bcde8c1b29 rtpmpvpay: fix timestamping of rtp buffers 
Incomming buffer is only pushed on the adapter at the end of the
handle_buffer function. But duration/timestamp of this buffer is already
taken into account for the current data in the adapter. This leads to
wrong rtp timestamps and extra latency.
 2010-10-12 15:17:02 +02:00
Sebastian Dröge
96fb89f86c examples: Fix build with GTK+ 3.0 2010-10-12 11:37:40 +02:00
Wim Taymans
ee7207aa3e rtspsrc: mark as a source 
Mark the rtspsrc element as a source.
Requires 0.10.31.1 now
 2010-10-11 15:12:51 +02:00
Sebastian Dröge
bcb4f50323 autodetect: Set GST_ELEMENT_IS_SOURCE flag on sources 2010-10-11 14:24:52 +02:00
Sebastian Dröge
5c1f6c890b switchsrc: Set the GST_ELEMENT_IS_SOURCE flag 2010-10-11 14:24:52 +02:00
Sebastian Dröge
72e05e498b configure: Require core 0.10.30.1 2010-10-11 14:24:52 +02:00
Zaheer Abbas Merali
f012ab67cc matroskamux: set offsets on outgoing buffers 2010-10-10 14:44:43 +01:00
IOhannes m zmölnig
b37845dac0 v4l2sink: Only get/set overlay params if needed 
it's perfectly ok for a video output device to not have overlay capabilities.
this patch removes the need to get/set the overlay parameters if the user
does not explicitely request one of the overlay properties
 2010-10-10 11:23:39 +02:00
IOhannes m zmölnig
4ba93e9f1a v4l2sink: Protect against NULL-pointer access 
gst_v4l2sink_change_state() would free the pool without checking whether there
was a valid pool...
 2010-10-10 11:23:39 +02:00
David Schleef
0aace5a0f3 Automatic update of common submodule 
From c4a8adc to 5a668bf
 2010-10-08 12:43:51 -07:00
Sebastian Dröge
6f5ce805fc Automatic update of common submodule 
From 5e3c9bf to c4a8adc
 2010-10-08 12:53:33 +02:00
Robert Swain
6a6f90e745 deinterlace: Fix required fields logic 
Both history_count and fields_required count from 1. As per the while loop
condition that follows this code, to perform the deinterlacing method, we need
history_count >= fields_required fields in the history. Therefore if we have
history_count < fields_required (not fields_required + 1), we need more fields.
 2010-10-06 15:05:36 +02:00
Andoni Morales Alastruey
0bffae750b flvmux: resend onMetada tag when tags changes in streamable mode 2010-10-06 09:14:24 +02:00
Arun Raghavan
4a244e0d55 qtdemux: AAC codec_data can be > 2 bytes long 
This fixes the assumption that DecoderSpecificInfo must be 2 bytes long
for AAC files. The specification allows HE-AAC to be explicitly
signalled in a backward compatible way. This is done by means of an
additional information after the regular AAC header. It is expected that
decoders that can play AAC but not HE-AAC will parse the header normally
and ignore extended bits, much as they do for the HE-AAC specific payload
in the actual stream.

https://bugzilla.gnome.org/show_bug.cgi?id=612313
 2010-10-05 19:45:31 +01:00
Mark Nauwelaerts
bb9a8a9b7d matroskademux: only unref buffer when no longer needed for cluster scanning 
Fixes #629047.
 2010-10-05 16:03:10 +02:00
Mark Nauwelaerts
e0d11f0644 matroskademux: avoid infinite cluster scanning 2010-10-05 16:03:08 +02:00
Wim Taymans
49d5d8f69e goom: take duration into account when doing QoS 
Take the duration of the frames into account so that we don't drop frames that
are only partially past the QoS deadline.
 2010-10-05 12:23:15 +02:00
Wim Taymans
cd5f31f751 goom: use adapter for timestamping 
Use the adapter timestamp code to get more accurate timestamps.
Fix latency calculation, we add our own latency in the worst case.
 2010-10-05 12:23:15 +02:00
Edward Hervey
9481c8a1a0 raw1394: Don't compile hdv1394src if libiec61883 isn't available 
Fixes #629896
 2010-10-04 22:31:32 +02:00
Andoni Morales Alastruey
4c2f5333bb icydemux: forward tag events 
https://bugzilla.gnome.org/show_bug.cgi?id=630205
 2010-10-04 20:10:02 +02:00
Wim Taymans
a060a65786 goom2k1: report our latency correctly 
Fixes #631303
 2010-10-04 19:01:03 +02:00
Wim Taymans
8c910a25fe goom2k1: add defines for default width/height/fps 
Add some defines for the default width/height/fps instead of using different
values in different places.
 2010-10-04 18:56:15 +02:00
Wim Taymans
418eae3ee3 goom: add latency compensation code. 
Implement a latency query and report how much latency we will add to the
stream.
Alse make some defaults for the default width/height/framerate

Fixes #631303
 2010-10-04 18:52:14 +02:00
Wim Taymans
1e310bc1ee test: add python version of the audio sender 
Add a python version of the audio sender pipeline.

Ported by Sp4rc on IRC.
 2010-10-04 17:56:57 +02:00
Wim Taymans
b50ce27b14 tests: Add python RTP client example 
Add a python version of the PCMA client app.

Ported by Sp4rc on IRC.
 2010-10-04 17:52:22 +02:00
Sebastian Dröge
a4c27169b6 rtp: Fix unitialized compiler warnings on OS X build bot 
These warnings are wrong though, the variables are only used in
the cases where they *are* initialized by the bit reader.
 2010-10-04 09:39:59 +02:00
Sebastian Dröge
c1877deee0 rtpg722pay: Fix uninitialized variable compiler warning 
The clock rate is always 8000 Hz according to the RFC and
the sampling rate must always be 16000 Hz.
 2010-10-03 23:49:08 +02:00
Vladimir Eremeev
8bf7381385 rtpjitterbuffer: improve article reference in comment block 
https://bugzilla.gnome.org/show_bug.cgi?id=631082
 2010-10-01 18:07:03 +01:00
Arun Raghavan
c65305d70e qtdemux: Use pbutils for H.264 profile/level extraction 
The functions used to extract this data have been moved to gstpbutils to
facilitate reuse.

https://bugzilla.gnome.org/show_bug.cgi?id=617318
 2010-10-01 17:07:36 +01:00
Arun Raghavan
9e786de9c8 matroskademux: Use pbutils for H.264 profile/level extraction 
The functions used to extract this data have been moved to gstpbutils to
facilitate reuse.

https://bugzilla.gnome.org/show_bug.cgi?id=617318
 2010-10-01 16:58:46 +01:00
Arun Raghavan
1a37a62454 qtdemux: Export MPEG-4 video profile and level in stream caps 
This uses gstpbutils to extract the profile and level from the video
object sequence and adds this to stream caps. This can be used as
metadata and for fine-grained decoder selection.

https://bugzilla.gnome.org/show_bug.cgi?id=616521
 2010-10-01 14:41:44 +01:00
Tim-Philipp Müller
64753bdbe8 qtdemux: fix aac channel override based on codec data for 7.1 case 2010-10-01 11:42:15 +01:00
Arun Raghavan
845a3d6c3d qtdemux: Export AAC profile and level in caps 
This exports the AAC profile and level in caps for use as metadata and
(eventually) for more fine-grained selection of decoders at
caps-negotiation time. (Doesn't work for HE-AAC yet though.)

https://bugzilla.gnome.org/show_bug.cgi?id=612313
 2010-10-01 11:41:53 +01:00
Wim Taymans
78e4a260b4 rtp: add G722 pay and depayloader 2010-09-30 18:34:36 +02:00
Thijs Vermeir
2c2c90a723 rtpjitterbuffer: update link to documentation 2010-09-30 12:08:49 +02:00
Thijs Vermeir
9c429de37a examples: fix indentation on rtp client example 2010-09-30 11:38:38 +02:00
Thijs Vermeir
92a1adfde8 examples: fix typo in port of rtp examples 2010-09-30 11:38:38 +02:00
Tim-Philipp Müller
a461b94629 wavenc: miniscule code clean-up 
GST_CLOCK_TIME_NONE is not something that should be used in connection with
GST_FORMAT_BYTES.
 2010-09-29 18:53:26 +01:00
Mark Nauwelaerts
8d69663026 avidemux: reverse playback; prevent overlap of subsequent fragments 2010-09-29 11:00:08 +02:00
René Stadler
0cfe24d132 rtspsrc: fix missing null-terminator in protocols array 
Fixes random crash regression from commit ae84ae.
 2010-09-28 16:21:48 +03:00
Wim Taymans
ef29a59903 rtspsrc: don't add /UDP in the transport, it's the default 
don't add the default UDP lower-transport, some servers don't seem to like it.

Fixes #630500
 2010-09-24 16:26:20 +02:00