Commit graph

7826 commits

Author SHA1 Message Date
Sebastian Dröge
7fec6843c0 audiowsinc{limit,band}: Allow much larger filter lengths now 2009-12-15 18:12:47 +01:00
Sebastian Dröge
119a6ce637 audiofxbasefirfilter: Fix frequency response calculation 2009-12-15 18:12:47 +01:00
Sebastian Dröge
8695581751 audiofxbasefirfilter: Remove dead assignments 2009-12-15 18:12:46 +01:00
Sebastian Dröge
cd2b1c1b58 audiofxbasefirfilter: Add special processing functions for Mono/Stereo
This provides another 7% speedup for the time domain convolution and 1.5%
speedup for the FFT convolution on Mono input.

This optimization assumes that the compiler simplifies calculations
and conditions on constant numbers and unrolls loops with a constant
number of repeats.
2009-12-15 18:12:46 +01:00
Sebastian Dröge
a3d7321c50 audiofxbasefirfilter: Add a "low-latency" mode
This will always use time-domain convolution, which lowers the latency.
With FFT convolution it's always a multiple of the kernel length,
with time domain convolution it's only the pre-latency of the filter kernel.
2009-12-15 18:12:46 +01:00
Sebastian Dröge
ca568ff079 audiofxbasefirfilter: Remove obsolete TODO comments 2009-12-15 18:12:46 +01:00
Sebastian Dröge
45edc1bbd8 audiofxbasefirfilter: Use samples everywhere instead of samples*channels sometimes 2009-12-15 18:12:46 +01:00
Sebastian Dröge
02960383c1 audiofxbasefirfilter: FFT convolution implementation
This provides a great speedup, especially the relationship between kernel
length and processing size is now logarithmic instead of linear. Below a
kernel size of 32 it's a bit slower, afterwards it's much faster:

17     0.788000 -> 0.950000
33     1.208000 -> 1.146000
65     2.166000 -> 1.146000
...
4097 107.444000 -> 1.508000

For sizes smaller 32 the normal time-domain convolution is chosen,
for larger sizes the FFT convolution is automatically used.

Fixes bug #594381.
2009-12-15 18:12:46 +01:00
Sebastian Dröge
ddafc20b28 audiofxbasefirfilter: Make most code parts independent of the processing functions and used convolution algorithm
Only remaining part is the residue pushing, which will be fixed later.
2009-12-15 18:12:46 +01:00
Sebastian Dröge
43576fb0cf audiofxbasefirfilter: Optimize time-domain convolution
Remove some redundant calculations, move comparisions out of
inner loops, etc.

This makes the convolution about 3 (!) times faster but
processing time is of course still proportional to the
filter size.
2009-12-15 18:12:46 +01:00
Sebastian Dröge
c5f955a3b6 audiofxbasefirfilter: Use _CAST macros in some places and do some calculations only once 2009-12-15 18:12:46 +01:00
Sebastian Dröge
abb437454e audiofxbasefirfilter: Rewrite timestamp tracking
It's much simpler now and doesn't introduce accumulating rounding
errors.
2009-12-15 18:12:46 +01:00
Sebastian Dröge
c57be62881 audiofxbasefirfilter: Rename some variables and change comments 2009-12-15 18:12:45 +01:00
Sebastian Dröge
742a7c7f50 audiofxbasefirfilter: Add const qualifier to the source data array 2009-12-15 18:12:45 +01:00
Sebastian Dröge
061ededa36 videomixer: Add MMX implementations of the AYUV blending and color filling functions
This provides a 20% speedup for blending and 100% for color filling.

The blending can probably be optimized even more.
2009-12-15 12:30:21 +01:00
Tim-Philipp Müller
d3a9f07669 id3demux: prefer two letter ISO 639-1 code for extended comment 2009-12-13 13:19:43 +00:00
Tim-Philipp Müller
6c4c8f8670 qtdemux: fix up language code extraction some more
Quicktime uses ISO 639-2 for language codes, but GST_TAG_LANGUAGE
is supposed to hold a ISO 639-1 code, so convert as needed using
the new API from -base.

See #602126.
2009-12-13 13:10:12 +00:00
Tim-Philipp Müller
b66f914586 matroska: fix language code writing and extraction
Matroska uses three-letter ISO 639-2B codes, but GST_TAG_LANGUAGE is
supposed to contain two-letter ISO 639-1 codes, so use new language
code mapping functions in -base to convert between those two as
needed.

Fixes #505823.
2009-12-13 12:51:13 +00:00
Tim-Philipp Müller
1b786258c2 avidemux: minor debug message changes
Fix up a few debug messages so that it's clearer what they mean.
2009-12-13 12:51:13 +00:00
Thiago Santos
52177fa056 Revert "qtdemux: Correctly parse classification tags"
This reverts commit cd883aa60c.

Previous code was correct, 4 is due to table and language code,
not only language code
2009-12-12 17:44:04 -03:00
Thiago Santos
cd883aa60c qtdemux: Correctly parse classification tags
In clsf atoms, the language code is 2 bytes long, not 4.
2009-12-12 16:31:35 -03:00
Sebastian Dröge
66d3ac8fb7 videomixer: Dequeue current buffer on FLUSH_STOP and don't unref NULL buffers
... NULL buffers shouldn't really happen anymore when popping the
buffer from GstCollectPads but better check for this and print a warning.
2009-12-12 16:55:13 +01:00
Sebastian Dröge
760eaf7b2a videomixer: Fix stupid mistake in last commit 2009-12-11 13:11:12 +01:00
Sebastian Dröge
089d9d9dba videomixer: Don't do floating point math in the inner processing loop for I420 blending 2009-12-11 12:36:42 +01:00
Wim Taymans
b8c2ccce4e rtspsrc: handle NULL and empty transport strings
When an RTSP extension returns NULL or an empty transport string, just ignore it
and try to get the next possible transport. Fixes playback of RealMedia streams.
2009-12-10 18:45:55 +01:00
Wim Taymans
6a44d8e198 rtspsrc: install event function on internal RTCP pad
Install a custom event function on the internal RTCP pad so that we can reply
TRUE to a latency event.
2009-12-10 18:45:55 +01:00
Sebastian Dröge
6f51dfba95 videomixer: Remove wrong comments, copied from the I420 blend function 2009-12-10 10:48:49 +01:00
Sebastian Dröge
93089ef445 videomixer: The queued duration is a signed integer
...and it will really be negative sometimes.
2009-12-09 21:15:07 +01:00
Sebastian Dröge
7418dee253 videomixer: Only pop buffers from collectpads after they're fully consumed
This decreases latency and memory usage because new buffers are only
accepted by collectpads if there's no queued buffer.
2009-12-09 21:03:57 +01:00
Sebastian Dröge
cd888c0531 matroskademux: Clean up position/duration handling
Also use the last end time for closing the segment, not the
start time of the last buffer.
2009-12-09 20:42:44 +01:00
Sebastian Dröge
0766a54138 matroskademux: Close the segment on EOS if the real duration is known 2009-12-09 16:50:02 +01:00
Sebastian Dröge
5ca96043ff matroskademux: Update duration if current buffer is already after the old duration 2009-12-09 16:46:18 +01:00
Sebastian Dröge
c9b1ab53fe matroskademux: Drop buffers that are after segment stop
...and if this happened for all streams go EOS.
2009-12-09 16:43:41 +01:00
Sebastian Dröge
276a61ab2a matroskademux: Fix position tracking and sending of filler segments 2009-12-09 16:41:04 +01:00
Sebastian Dröge
b0f8978fd8 videomixer: Use gst_util_uint64_scale_int() for fps to seconds per frame calculations 2009-12-09 16:15:09 +01:00
Sebastian Dröge
3ddb75e3c5 matroskademux: Keep the segment stop position for update newsegment events 2009-12-08 17:34:15 +01:00
Sebastian Dröge
648f0f0b50 ladspa: Remove the sources from gst-plugins-good
It's disabled anyway and the latest version of it is in
gst-plugins-bad. Fixes bug #603779.
2009-12-04 14:43:41 +01:00
Wim Taymans
ee6d7fd2db avidemux: init current_entry in push mode
Set the current_entry to 0 (instead of -1) in push mode so that we correctly
calculate the current frame number and timestamp.

Add some more debug info and fic the duration debug.
2009-12-04 13:52:49 +01:00
Tim-Philipp Müller
24b93d82ec rtspsrc: fix major memory leak when playing back rtsp video streams
Don't forget to unref QoS, navigation and latency events when
dropping them.
2009-12-04 11:14:03 +00:00
Tim-Philipp Müller
d0b25845ec matroskademux: only send pending tags with newsegment events
Send pending tags only from the streaming thread, just after we've sent
the newsegment event, not with e.g. flush-start. This not only does the
right thing, but also makes sure we're not trampling over variables set
up in the streaming thread from the seeking thread in case someone tries
to issue a seek just as the demuxer is parsing the headers.

Fixes #601617. Spotted by Ognyan Tonchev.
2009-12-04 11:13:31 +00:00
Thiago Santos
ff4ac9ddf6 qtdemux: fix debug message printf args
Fixes debug message printf format to make it build in mac's gcc
2009-12-03 17:49:55 -03:00
Thiago Santos
8cb3452ac4 shout2: Convert delay correctly
Use GST_MSECOND to convert delay in msecs to nanosecs

Fixes #603547
2009-12-02 13:33:20 -03:00
Wim Taymans
384c4a2495 jpegdec: reset segment info after flush
Reset the segment info after a flush. We use the segment for handling QoS and if
we don't reset the segment, QoS is basically disabled after a flushing seek.
2009-12-01 19:25:04 +01:00
Tim-Philipp Müller
0e777bf759 Automatic update of common submodule
From 87bf428 to 47cb23a
2009-12-01 15:07:06 +00:00
Sebastian Dröge
5ed53fb4ac Automatic update of common submodule
From da4c75c to 87bf428
2009-12-01 14:15:46 +01:00
Aurelien Grimaud
07f27f0efd rtpsession: avoid buffer ref/unref pairs for CSRCs
We ref the buffer before pushing it downstream in order to get the CSRCs of it
after pushing. This causes performance problems when downstream elements want to
change the metadata because the buffer needs to be subbuffered.

Instead, read and store the CSRCs of the buffer in an array before pushing it
and process the array after pushing the buffer. This allows us to remove the
ref/unref pair.

Fixes #603376
2009-11-30 15:59:50 +01:00
Wim Taymans
4519bdc983 shout2: use gstpoll for timeouts
Use our own GstPoll based timeout instead of the shout sleep so that we can
interrupt when doing a state change and shutting down.

Fixes #602887
2009-11-28 19:23:26 +01:00
Wim Taymans
5c8527bb20 check: fix jitterbuffer check
Make sure we set a base_time on the element.
Fix the timeout to at least twice the jitterbuffer latency.
Enable previously failing tests.
Remove impossible checks.
2009-11-28 12:25:06 +01:00
Edward Hervey
2c694abd8e Automatic update of common submodule
From 53a2485 to da4c75c
2009-11-27 18:55:20 +01:00
Mark Nauwelaerts
e49e71a1d9 rtph264depay: optionally merge NALUs into Access Units
... which may be expected/desired by some downstream decoders
(and spec-wise highly recommended for at least non-bytestream mode).
2009-11-26 17:29:26 +01:00