Commit graph

7267 commits

Author SHA1 Message Date
Sebastian Dröge
403f353bba audio: Add gobject-introspection support 2009-09-05 11:09:33 +02:00
Sebastian Dröge
61ae0059a4 app: Add gobject-introspection support 2009-09-05 11:09:28 +02:00
Sebastian Dröge
78063ab4be Automatic update of common submodule
From 00a859e to 19fa4f3
2009-09-05 10:20:24 +02:00
Wim Taymans
f1b209f409 typefind: fix midi typefinding
We already have a audio/midi typefinder so don't override it with the midi in
RIFF typefinder or else we fail to detect plain midi files.
2009-09-04 15:48:06 +02:00
Wim Taymans
9e83339cf6 uridecodebin: do buffering for more uris
Add ssh://, ftp://, sftp://, myth:// to the list of uris that require
buffering.

Fixes #594020
2009-09-04 11:33:04 +02:00
Sebastian Dröge
a69ffb5886 typefindfunctions: Add typefinder for Midi inside RIFF
This is a standard Midi file format that should be supported by
all Midi decoders and also has the mimetype audio/mid according to
the Midi specification homepage.

Fixes bug #594094.
2009-09-04 07:36:10 +02:00
Wim Taymans
7a7663476f audiortppay: add some debugging 2009-09-03 18:53:19 +02:00
Wim Taymans
c1db9ebb20 audiortppay: handle gaps
Add various conversion functions between time<->bytes<->rtptime that will be
used later on.
Refactor the min/max packet length code so that it can be used for both
sample/frame based payloaders. Cache the returned values.
code cleanups.
When we discover a DISCONT buffer, make the outgoing RTP timestamps have the
same gap as the GStreamer timestamps gap.
2009-09-03 17:59:00 +02:00
Wim Taymans
3a3c6f309c audiortppay: fix frame duration calculations
Fix the calculation of the frame duration and rtp timestamps.
Add some debugging
2009-09-03 17:59:00 +02:00
Wim Taymans
bfc19462bb rtppay: add some debugging 2009-09-03 17:59:00 +02:00
Wim Taymans
bb91a7b47c audiortppay: use offsets for RTP timestamps
Have a custom sample/frame function to generate an offset that the base class
will use for generating RTP timestamps. This results in perfect RTP timestamps
on the output buffers.
Refactor setting metadata on output buffers.
Add some more functionality to _flush().
Handle DISCONT on the input buffers and set the marker bit and DISCONT flag on
the next outgoing buffer.
Flush the pending data on EOS.
2009-09-03 17:58:59 +02:00
Wim Taymans
c1ae0a2003 audiortppay: move function around 2009-09-03 17:58:59 +02:00
Wim Taymans
5808041f44 audiortppay: fix sample duration calculation 2009-09-03 17:58:59 +02:00
Wim Taymans
299ab7be0e audiortppay: more refactoring
Unify the sample/frame buffer handling code by making the functions plugable.
2009-09-03 17:58:59 +02:00
Wim Taymans
fb5037f727 audiortppayload: refactor some more
Refactor getting the packet min/max size and alignment code.
Refactor converting bytes to time.
change some variable to something shorter.
2009-09-03 17:58:59 +02:00
Wim Taymans
1c6b71af03 audiortppayload: refactor and cleanup
Always use the adapter when we need to fragment the incomming buffer. Use more
modern adapter functions to avoid malloc and memcpy. The overall result is that
the code looks cleaner while it should be equally fast and in some case avoid a
memcpy and malloc.
Use the adapter timestamping functions for more precise timestamps in case of
weird disconts.
Cache some values instead of recalculating them.
Add gst_base_rtp_audio_payload_flush() to flush a certain amount of bytes from
the internal adapter.

API: GstBaseRTPAudioPayload::gst_base_rtp_audio_payload_flush()
2009-09-03 17:58:59 +02:00
Tim-Philipp Müller
432612b035 Update common 2009-09-03 16:56:55 +01:00
Wim Taymans
50b9640d01 basertppay: add property to disable perfect RTP time
Add a property to disable the generation of perfect RTP timestamps. By default
it is active.

API: GstBaseRTPPayload::perfect-rtptime
2009-09-03 11:29:23 +02:00
Wim Taymans
3a4edea56d basertppay: allow subclasses to influence RTP time
Allow subclasses to use the OFFSET field on RTP buffers to influence the way in
which RTP timestamps are generated. Usually timestamps are created from the
GStreamer timestamps on the buffer, which could result in imperfect RTP
timestamps.
2009-09-03 11:15:20 +02:00
Wim Taymans
5a479669d4 basertppay: add macro to cast 2009-09-03 11:15:20 +02:00
Wim Taymans
bc3c8a1564 audiopayload: code cleanups 2009-09-03 11:15:20 +02:00
Wim Taymans
3c29efa692 audiortppayload: don't check adapter
the adapter is never NULL so we don't need to check it.
Use _scale functions to avoid overflows.
2009-09-03 11:15:20 +02:00
Tim-Philipp Müller
4cacc441d8 typefinding: move gio-based xdg mime typefinder from -bad to -base
Its purposes is mainly to avoid false positives (e.g. mp3 typefinder
reporting a 20% probability and somesuch). Won't be registered if
the gio plugin has been disabled via ./configure --disable-gio.
2009-09-03 09:01:47 +01:00
Tim-Philipp Müller
889c318798 subparse: GstAdapter is not a GstObject and should be freed with g_object_unref 2009-09-01 15:16:17 +01:00
Tim-Philipp Müller
95af14d7bc v4lsrc: fix timestamping for when we do not have a clock yet
Should fix #559049.
2009-09-01 15:16:06 +01:00
Tim-Philipp Müller
e2e5a1da06 v4lsrc: don't log not-yet-initialised integer value 2009-09-01 15:08:18 +01:00
Tim-Philipp Müller
1af601d403 v4lsrc: avoid unnecessary run-time type checks in custom buffer finalize
And reflow code to be more indent friendly.
2009-09-01 15:08:10 +01:00
Jonas Holmberg
ec91d508af basertppayload: Make instance init faster by not reading /dev/urandom 3 times
... which is the default seed when creating a new GRand. Because
GLib in older versions used buffered IO this would take a lot of time.

Instead use the global GRand for getting random numbers and keep the
three instance GRand for backward compatibility with a simple seed.

Fixes bug #593284.
2009-09-01 10:39:52 +02:00
Stefan Kost
e7368354d5 adder: improve caps filter functionality. Fixes #590146.
Also use the capsfilter if there is no src-peer as the caps constrain what
we can do. Don't create any_caps as a default, as we check for NULL to skip the
filtering. This is a (small) performance regression as we always intersect
otherwise.
2009-08-31 22:48:01 +03:00
Sebastian Dröge
2194166e05 decodebin2: Post missing plugin messages before any error messages 2009-08-31 11:10:55 +02:00
Wim Taymans
008c760b6b cddabasesrc: safely handle the indexes 2009-08-28 19:06:57 +02:00
Wim Taymans
cbc6c0dfea def: add new rtsp symbols 2009-08-28 19:06:44 +02:00
Wim Taymans
e40b262ab7 basertppayload: whitespace fixes. 2009-08-28 14:09:02 +02:00
Marc-André Lureau
605f3c2942 Bug 593035 - set IN_CAPS for streamheader buffer 2009-08-28 08:44:55 +01:00
Sebastian Dröge
460dc94d23 playbin: The internally linked pad of the selector might be NULL in some cases 2009-08-26 16:56:19 +02:00
Sebastian Dröge
67a0ef9b3a playbin: Fix iterate internal linked pads functions for the stream selectors
This now used the new gst_iterator_new_single() function and as a side effect
fixes bug #592864.
2009-08-26 16:45:49 +02:00
Sebastian Dröge
72f3587f04 riff: Add support for AVF files
AVF is valid RIFF but has AVF0 has first fourcc instead of RIFF.

Fixes bug #593117.
2009-08-26 09:10:19 +02:00
Sebastian Dröge
3c8ff21ba2 typefindfunctions: Detect AVF files as RIFF files too
AVF is valid RIFF but has AVF0 as first fourcc instead of RIFF.

Partially fixes bug #593117.
2009-08-26 09:10:19 +02:00
Sebastian Dröge
e22c843d0e audioresample: Add unit test for checking for timestamp drifts
This also checks for perfect timestamping and offsetting.
2009-08-26 09:10:18 +02:00
Sebastian Dröge
1e450f21f8 audioresample: Fix drain processing
In case we have to convert internally don't process output length input samples
but history length input samples.
2009-08-26 09:10:18 +02:00
Sebastian Dröge
01408497a1 audioresample: Improve debugging a bit in the unit test 2009-08-26 09:10:18 +02:00
Sebastian Dröge
2e585ac7ac audioresample: On the first buffer we need discont handling
Otherwise we won't get upstream timestamps and everything and all
output buffers would have -1 timestamps.
2009-08-26 09:10:18 +02:00
Руслан Ижбулатов
5d96fd4bf1 subparse: Remove dependency on regex.h as it's not used anyway
Fixes bug #592544.
2009-08-26 09:10:17 +02:00
Kipp Cannon
86b4c51c8c audioresample: Fix buffer overflow when pushing the drain 2009-08-26 09:10:17 +02:00
Kipp Cannon
a69068d70d audioresample: Fix timestamp drift
Fixes bug #591934.
2009-08-26 09:10:17 +02:00
David Schleef
0e9bc5125a Remove Ronald Bultje from Authors field
Replaced with "GStreamer maintainers
<gstreamer-devel@lists.sourceforge.net>" or just removed,
depending on the number of other authors.
2009-08-24 11:37:01 -07:00
Wim Taymans
c3ebeec5a5 playbin2: fix refcounting of _get_sink()
g_value_set_object() increases the refcount of the sink, which is not needed
because the object should already be refcounted. Make sure this is always the
case and use g_value_take_object().

Fixes: #592884
2009-08-24 15:08:36 +02:00
Peter Kjellerstedt
8ce3612b71 rtsp: Mark Transport as supporting multiple values. 2009-08-24 14:39:16 +02:00
Peter Kjellerstedt
2882c22d95 rtsp: Added missing Since tags. 2009-08-24 13:58:50 +02:00
Eero Nurkkala
8ad8591e41 ringbuffer: Improve audiosink startup performance
When we start the ringbuffer, immediatly continue processing samples if the
writer prepared some for us.

Fixes #545807
2009-08-24 13:30:11 +02:00