Commit graph

4063 commits

Author SHA1 Message Date
Wim Taymans
afc3c674c0 avidemux: skip RIFF and index in push mode
When we are in push mode, we can encounter RIFF and idx tags in the data chunk
when we are dealing with ODML files. In these cases, simply skip the chunks and
continue streaming instead of going EOS.
2010-01-20 11:47:04 +01:00
Wim Taymans
570319822a avidemux: more DISCONT handling
Add some debug in the DISCONT handling code.
When we receive a DISCONT in push mode, mark all streams as DISCONT.
2010-01-20 11:47:04 +01:00
Wim Taymans
40e3b0189a avidemux: reset on flush events
When we receive a flush event on the sinkpad, reset the EOS state and the
flowreturn of all streams. Also mark the streams with a DISCONT.
2010-01-20 11:47:03 +01:00
Wim Taymans
183d450113 avidemux: rename some variable
Rename the seek_event variable to seg_event because it really contains the
newsegment event that needs to be pushed.
2010-01-20 11:47:03 +01:00
Olivier Crête
c4fa559f15 rtph264pay: Don't set profile-level-id in out caps
The profile-level-id represents restrictions on what can be sent, it does not
describe the stream. So it should be reflected in the sink caps of the
payloader, not the src caps.

https://bugzilla.gnome.org/show_bug.cgi?id=607353
2010-01-19 13:47:38 +01:00
Olivier Crête
7a0590b1f1 rtph264pay: Don't ignore the return value from set_outcaps
https://bugzilla.gnome.org/show_bug.cgi?id=607353
2010-01-19 13:35:37 +01:00
Sebastian Dröge
2261bd8346 deinterlace: Fix license and copyright headers 2010-01-18 17:44:17 +01:00
Wim Taymans
fb716a6250 avidemux: avoid some typecasting 2010-01-15 18:15:14 +01:00
Wim Taymans
592b440911 avidemux: avoid some type checks 2010-01-15 18:13:24 +01:00
Wim Taymans
d4301d900f avidemux: fallback to avih duration
when we have not yet parsed the indexes (in push mode, for example) use
the duration as given in the avih header instead of -1.
2010-01-15 18:09:15 +01:00
Thiago Santos
e61a71b490 qtdemux: g_free is NULL safe 2010-01-15 13:42:30 -03:00
Thiago Santos
b07f406634 qtdemux: use DEMUX errors, instead of DECODE
qtdemux should use DEMUX errors, and not DECODE

Conflicts:

	gst/qtdemux/qtdemux.c
2010-01-15 13:42:30 -03:00
Thiago Santos
b988ff4f57 qtdemux: Minor refactor
Replace repeated code with a function call
2010-01-15 13:42:30 -03:00
Thiago Santos
92a83e016a qtdemux: Handle another kind of redirect trak
Some traks might contain a redirect rtsp uri inside
hndl atom (which is a dref atom entry). This commit makes qtdemux
post a message when it finds one of these traks and there are
no other traks.

Fixes #597497
2010-01-15 13:42:29 -03:00
Thiago Santos
06de494640 qtdemux: Post error when reaching EOS without pads
Post an error when EOS is reached and there are no src pads
2010-01-15 13:42:22 -03:00
Thiago Santos
b53a45ed44 qtdemux: Do not post empty redirect messages
Some misinterpreted data could result in posting redirect messages
with empty redirect strings. It is better not to post them.

An example is the file on bug #597497
2010-01-15 13:13:59 -03:00
Mark Nauwelaerts
891ca1f4d3 matroskademux: polish last buffer end time usage
That is, reset it upon seek, and note that (rarely) last pushed buffer
time might precede segment start.
2010-01-14 18:19:25 +01:00
Stefan Kost
404e673ac0 videomixer: use 'q' constraint instead of 'r'
This avoids the "bad register name `%dil'" compilation errors on 32bit where
because of 'r' gcc puts the value in a general purpose register and then tries
to access the lower part as %dil/%sil which is not existing on 32bit. 'q' requests
a-d registers
2010-01-13 16:48:46 +02:00
Stefan Kost
7e3783cbac avi: add missing include for sscanf 2010-01-13 16:44:58 +02:00
Sebastian Dröge
4a0f441c59 equalizer: Fix property description for the 3rd band of the 10band equalizer
The frequency is actually 237 Hz, not 227 Hz.

Fixes bug #606692.
2010-01-13 09:36:03 +01:00
Kipp Cannon
d009678bc5 audioamplify: Allow negative amplifications
Fixes bug #606807.
2010-01-13 09:22:20 +01:00
Edward Hervey
3f5add8820 qtdemux: use G_GSIZE_FORMAT for platform independent gsize qualifier
Fixes build on macosx
2010-01-12 17:39:05 +01:00
Mark Nauwelaerts
59224d77f8 matroskademux: refactor eos sending when pausing loop
Also, prevent hanging if no pads yet on which to send eos by
posting a message instead.
2010-01-11 21:15:47 +01:00
Mark Nauwelaerts
ae515fead4 matroskademux: standardize seek handling
... which implies fixing some corner cases.
2010-01-11 21:15:46 +01:00
Mark Nauwelaerts
927c22bdc4 matroskamux: use more generic xiphN_streamheader_to_codecdata helper 2010-01-11 21:15:43 +01:00
Mark Nauwelaerts
847d1dd4ed matroskamux: reflow audio and video setcaps and improve logging
Also ensure width and height are available as they are mandatory
in matroska specs.
2010-01-11 21:15:41 +01:00
Michael Smith
144fbd2d8f qtdemux: fix offset for type 2 mp4a sound sample descriptions.
Allows us to correctly find the esds (and thus the codec data) for such
mp4a files.
2010-01-11 11:48:29 -08:00
Thiago Santos
fa32e08d91 rtpmp4g(de)pay: Only handle raw aac
rtpmp4g(de)pay should only handle raw AAC streams
2010-01-11 15:46:50 -03:00
Sebastian Dröge
daa52708b3 videomixer: Implement basic QoS
This drops frames if they're too late anyway before blending and all
that starts but QoS events are not forwarded upstream. In the future
the QoS events should be transformed somehow and forwarded upstream.
2010-01-11 19:32:35 +01:00
Thiago Santos
c563dd7eb2 rtpmp4a(de)pay: Only accept raw aac
rtpmp4a(de)pay should only handle raw aac to conform to the RFC
2010-01-11 15:00:00 -03:00
Sebastian Dröge
6158f401a1 videomixer: Add MMX implementations for I420 and all non-alpha RGB formats 2010-01-11 18:37:45 +01:00
Sebastian Dröge
2950262186 videomixer: Refactor processing functions
This allows easier plugging of optimized processing functions
in the future, like for SSE or AltiVec.
2010-01-11 18:37:44 +01:00
Thiago Santos
5975b01b01 avimux: matroskamux: rename aac's stream-format to raw
AAC's none stream-format has been renamed to raw, rename
on avimux and matroskamux as well
2010-01-11 13:26:32 -03:00
Thiago Santos
1314853210 matroskamux: Only accept raw aac
makes matroskamux reject aac streams that are not
in raw format (stream-format=none)

Fixes #598350
2010-01-11 12:32:29 -03:00
Thiago Santos
bacd350483 avimux: Only accept raw aac
makes avimux reject aac streams that are not
in raw format (stream-format=none)

Fixes #598350
2010-01-11 12:32:27 -03:00
Robert Swain
866d13e7b9 qtdemux: Oops. The gpointer cast is needed because of the const
qualifiers on the data elements
2010-01-11 10:38:10 +01:00
Robert Swain
4ac643c2d9 qtdemux: Debug -> info level for a message for benchmarking index parsing
The extra message output at higher levels affects the accuracy of the
benchmark.
2010-01-11 10:17:54 +01:00
Robert Swain
c93ea637ef qtdemux: Don't check for NULL pointers or cast to gpointer as this is
not needed
2010-01-11 10:05:10 +01:00
Robert Swain
a340359127 qtdemux: Refactor stbl sub-atom freeing. Free when index has been
completely parsed.
2010-01-11 09:50:33 +01:00
Robert Swain
3daf1871c1 qtdemux: Avoid whitespace commits due to inconsistent GNU indent
behaviour
2010-01-11 09:50:33 +01:00
Tim-Philipp Müller
e1bff64f00 qtdemux: remove newline at end of debug statement 2010-01-11 00:10:34 +00:00
Havard Graff
4ead3d85bf multiudpsink: Compiler warning fixes for Windows
Just simple missing casts

Fixes bug #606438.
2010-01-09 17:17:23 +01:00
Thiago Santos
8e84d457b2 avidemux: Use more glib and be safer
Be safer on sscanf by limiting string format sizes.
Remove useless parameter and use g_strndup.
2010-01-08 11:33:02 -03:00
Thiago Santos
c0e184641a avidemux: Simplifying code
Greatly simplify the IDIT chunk handling by using sscanf
instead of 'manually' parsing. Also replaces strncasecmp and
is_alpha/is_digit with glib versions.
2010-01-08 10:51:17 -03:00
Thiago Santos
7024ce14cf avidemux: it's feb for february
Fix typo in last commit.
2010-01-08 10:18:30 -03:00
Thiago Santos
a5197a94ee avidemux: Parse and post IDIT dates
Parses and post date tags contained in IDIT chunks.

Fixes #503582
2010-01-08 09:17:22 -03:00
Sebastian Dröge
a9a5e0c7e1 audiofxbasefirfilter: Add property for not draining the history on kernel changes
Currently this only works if the kernel size doesn't change, in the future
it will be possible to change the kernel size too without draining
the complete history and without loosing anything.

Partially based on a patch by
Thiago Santos <thiago.sousa.santos@collabora.co.uk>
2010-01-07 17:28:43 +01:00
Wim Taymans
ed22a97478 rtph264pay: remove weird memcmp code
Use plain memcmp for comparing memory instead of the custom buggy one.

Fixes #606198
2010-01-07 17:00:20 +01:00
Edward Hervey
3e08a0cb4e level: fix typo in 'message' property description 2010-01-07 15:38:36 +01:00
Wim Taymans
4c1947045e rtpg728pay: remove unused adapter peek 2010-01-06 13:45:59 +01:00
Michael Smith
7f442ab1c1 qtdemux: Add support for wave-style audio in qt.
Uses gstriff to parse the wave headers appropriately. Tested with MS-ADPCM
content.
2010-01-05 12:11:31 -08:00
Olivier Crête
63a9db5826 rtpg729pay: Simplify adapter usage
https://bugzilla.gnome.org/show_bug.cgi?id=606050
2010-01-05 13:23:26 -05:00
Olivier Crête
0a18587792 rtpg729pay: Support ptime from caps
https://bugzilla.gnome.org/show_bug.cgi?id=606050
2010-01-05 13:23:26 -05:00
Olivier Crête
321829f595 rtp: Add maxptime to the README
https://bugzilla.gnome.org/show_bug.cgi?id=606050
2010-01-05 13:23:26 -05:00
Wim Taymans
b32ddfc174 rtpg723depay: add G723 depayloader 2010-01-05 19:03:06 +01:00
Wim Taymans
ca7ecdf2f3 rtpg729depay: remove unused variable 2010-01-05 19:02:39 +01:00
Wim Taymans
d6d06630e8 rtpg723pay: rewrite payloader
Handle all 3 packet sizes according to RFC 3551.
Totally untested, we don't have a G723 encoder.

Fixes #605882
2010-01-05 18:33:25 +01:00
Wim Taymans
48615d5e98 qtdemux: fix chunk counter 2010-01-05 15:51:55 +01:00
Wim Taymans
17630760f4 qtdemux: more work at reducing loop overhead
Try to avoid derefs when parsing the index. Save the state into the structures
when we exit the loop instead of for each iteration.
2010-01-05 15:51:52 +01:00
Wim Taymans
91a5e5138f qtdemux: cleanups and make duration more accurate
Make the QtDemuxSample struct smaller by keeping the duration and the pts_offset
as their 32 bit values.
Make some macros to calculate PTS, DTS and duration of a sample.
Deref the sample index less often by keeping a ref to the sample we're dealing
with.
2010-01-05 15:51:50 +01:00
Wim Taymans
22eb18f828 qtdemux: simplify logic to calculate duration
Since we no longer store the timestamp and duration in nanoseconds, we can now
simply store the duration as-is.
2010-01-05 15:51:48 +01:00
Robert Swain
1c27ed4dae qtdemux: Store timestamps in mov format in the index
This allows faster building of the index upon seeks so that scaling of
timestamps only occurs when actually needed.
2010-01-05 15:51:45 +01:00
Wim Taymans
86021857c5 qtdemux: make seeking in push mode work
Move sample position checks into qtdemux_parse_samples where we can protect it
with a lock.
Refactor and make an qtdemux_ensure_index function.
Rename qtdemux_do_push_seek to qtdemux_seek_offset in order to avoid confusion
with gst_qtdemux_do_push_seek.
2010-01-05 15:51:43 +01:00
Wim Taymans
3b643817be qtdemux: move error code out of normal flow 2010-01-05 15:51:40 +01:00
Robert Swain
4b2b7067b6 qtdemux: Add push mode seek support for seeking to obtain the moov atom 2010-01-05 15:51:36 +01:00
Wim Taymans
8c5a822250 rtspsrc: fix on-npt-stop signal warnings for RDT
The RDT manager does not implement this signal so we need to check for it before
trying to connect to it.
2010-01-05 12:23:16 +01:00
Stefan Kost
fd9530d2d5 avimux: fix typo in warning message 2010-01-05 00:12:44 +02:00
Arun Raghavan
e9f9164fb6 qtdemux: Add tags for average and maximum bitrate
Fixes #599300.
2009-12-31 18:25:20 +00:00
Thiago Santos
173be1422c audiofxbasefirfilter: do not try to alloc really large buffers
When nsamples_out is larger than nsamples_in, using unsigned
ints lead to a overflow and the resulting value is wrong and
way too large for allocating a buffer. Use signed integers
and returning immediatelly when that happens.
2009-12-26 16:59:14 -03:00
Wim Taymans
362785df88 videomixer: optimize blend code some more
Use more efficient formula that uses less multiplies.
Reduce the amount of scalar code, use MMX to calculate the desired
alpha value.
Unroll and handle 2 pixels in one iteration for improved pairing.
2009-12-25 12:38:45 +01:00
Wim Taymans
4f9ded7742 videomixer: scale and clamp
Scale and clamp to the max alpha values.
2009-12-24 22:59:09 +01:00
Wim Taymans
0620797a18 alpha: scale and clamp alpha to its full extend
Convert the alpha value to 0->255 when setting and to 0->256 when using as
a scaling factor. This makes sure we can reach the full opacity value of 0xff in
all cases.
2009-12-24 22:50:31 +01:00
Wim Taymans
a65240d1c1 rtspsrc: fix some comments, remove property check
Fix some comments, clarify some FIXMEs
Remove the on-ntp-stop signal check now that the jitterbuffer is in
-good and we know that it supports this signal.
2009-12-24 22:23:01 +01:00
Wim Taymans
3c0f18d765 videomixer: some trivial cleanups 2009-12-24 21:45:12 +01:00
Thiago Santos
ac03ad782a rtspsrc: Parse all rtpinfo entries
Do not forget to parse all rtp-info entries, instead of
parsing the first one only.

Fixes #605222
2009-12-24 17:08:22 -03:00
Thiago Santos
5d86010dad qtdemux: perf tag should map to GST_TAG_ARTIST 2009-12-24 17:06:16 -03:00
Wim Taymans
fe529e71c5 interleave: fix weird indentation 2009-12-24 17:03:02 +01:00
Wim Taymans
59dc9dac03 rtph263ppay: use faster _adapter_copy() whem possible 2009-12-24 17:01:54 +01:00
Mark Nauwelaerts
05307c46e7 rtph264pay: fix uninitialized variable 2009-12-23 19:39:05 +01:00
Wim Taymans
9f098b352b rtp: use boilerplate 2009-12-23 13:09:54 +01:00
Wim Taymans
2ee7f58416 rtpL16pay: convert to baseaudiopayload
Use GstRTPBaseAudioPayload as the base class. This saves a lot of code and fixes
a bunch of problems that were already solved in the base class.

Fixes #853367
2009-12-23 00:38:05 +01:00
Wim Taymans
cdb8c718bb rtppcmapay: the boilerplate macro sets parent_class 2009-12-23 00:30:49 +01:00
Wim Taymans
05418f1687 rtpbin: avoid some structure copies
Don't make copied in the getter and setter for SDES in the RTPSource. This
avoids a couple of copies of the SDES structure when generating RTCP
packets.
2009-12-22 22:27:21 +01:00
Pascal Buhler
c3448f978e rtpmanager: improve SDES handling
Store SDES internally as a struct to support multiple PRIV values.
Include all values set in SDES struct when sending RTCP SDES.
2009-12-22 21:43:25 +01:00
Wim Taymans
251401aef1 rtph263depay: add some fixmes 2009-12-22 14:41:35 +01:00
Wim Taymans
564581e1b8 rtph263depay: baseclass handles timestamps for us 2009-12-22 14:35:13 +01:00
Wim Taymans
27ff4a8a47 rtph263depay: reset start variable properly 2009-12-22 14:27:40 +01:00
Marco Ballesio
74b3439374 Drop the whole frame if a packet is lost.
Fixes #582575
2009-12-22 11:48:52 +01:00
Wim Taymans
4687199348 rtph264pay: add option to insert PPS/SPS in streams
Add a new spspps-interval property to instruct the payloader to insert
SPS and PPS at periodic intervals in the stream.
Rework the SPS/PPS handling so that bytestream and AVC sample code both use the
same code paths to handle sprop-parameter-sets. This also allows to have the AVC
code to insert SPS/PPS like the bytestream code.

Fixes #604913
2009-12-21 20:45:54 +01:00
Jonathan Conder
1112090589 qtdemux: Adds new tags
Adds some new tags mapping to qtdemux.

Fixes #599759
2009-12-21 12:03:30 -03:00
Wim Taymans
9734699788 rtpbin: add property to remove pads automatically
Add a property called autoremove to automatically remove the pads of sources
that timed out.

Fixes #554839
2009-12-21 15:07:44 +01:00
Wim Taymans
c611bbaa8e ssrcdemux: fix comparison
A NULL means no pad was found.
2009-12-21 15:07:34 +01:00
Michael Smith
eab08d67b3 multiudpsink: pass length parameter to g_convert 2009-12-20 17:26:15 -08:00
Edward Hervey
188725811f matroska: Fix unitialized variable.
Yes, it's stupid, but macosx compilers are even more stupid.
2009-12-18 12:46:06 +01:00
Sebastian Dröge
3ac6f5e48b videomixer: Fix assembly compilation on x86
Fixes bug #604814.
2009-12-17 18:14:55 +01:00
Branko Čibej
7b107f64f3 rganalysis: fix timestamp rounding
Use scaling function to round and avoid overflows.

Fixes #604352
2009-12-17 17:37:03 +01:00
Tiago Katcipis
908a9ee63b rtp: add G723 payloader
Fixes #597823
2009-12-17 17:27:42 +01:00
Wim Taymans
cc277b4a26 qtdemux: Fix ALAC codec_data parsing
Fixes #604611
2009-12-17 16:23:56 +01:00
Thiago Santos
4063bb87e8 qtdemux: Remove cpp style coments
Removes // comments and replace them with /* */ comments
2009-12-16 17:28:30 -03:00
Mark Nauwelaerts
c9a0d2339e matroskademux: also consider BlockNumber indicated in index when seeking 2009-12-16 12:48:02 +01:00
Mark Nauwelaerts
900ff7247e matroskademux: support push based mode
Fixes #598610.
2009-12-16 12:46:40 +01:00
Mark Nauwelaerts
e4183c6904 matroskademux: fix ebml read cache usage 2009-12-16 12:46:37 +01:00
Sebastian Dröge
0a0f7ecc16 videomixer: Use movzbl instead of movzxb for moving one byte to a l register
For some reason latest gcc/binutils accept movzxb here while
movzbl would be correct and is the only thing accepted by older
gcc/binutils.

Fixes bug #604679.
2009-12-16 10:50:32 +01:00
Sebastian Dröge
9e45038d8d videomixer: src/dest are input and output of the AYUV blending MMX assembler 2009-12-16 06:59:01 +01:00
Sebastian Dröge
c26ccb9722 audiowsincband: Use the same upper length limit as audiowsinclimit 2009-12-15 18:18:54 +01:00
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
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
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
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
Mark Nauwelaerts
baa28ddedf qtdemux: fix timestamp datatype 2009-11-26 17:29:03 +01:00
Wim Taymans
8070ae967b jitterbuffer: avoid using wrong clock-rate
Check for a valid clock-rate before attempting to estimate the npt
stop time.
2009-11-25 10:38:23 -06:00
Wim Taymans
5682e2bf01 rtpbin: fix typo in comments 2009-11-25 10:37:30 -06:00
Michael Smith
9d6adc8f3c multiudpsink: return error message on windows too. 2009-11-24 11:13:06 -08:00
Michael Smith
d4826d987c multiudpsink: first phase of fixing up error reporting for windows. 2009-11-24 10:58:49 -08:00
Thiago Santos
b59dc3e5fb avimux: also set the suggested buf size for audio
We were only setting the suggested buf size for video,
we can set it for audio as well.

This and 195e14529d80ef318ce3a778c1995efb11f266cd
fix an issue that prevented seeking on large avi files
on WMP (non-recent versions).
2009-11-24 12:44:57 -03:00
Thiago Santos
831b1e958a avimux: fix indx duration for PCM audio
GstBuffers for PCM audio usually contains more than
1 sample, we need to get the total number of samples to set
the indx duration.
2009-11-24 12:44:56 -03:00
Thiago Santos
8dd78015f1 avimux: Audio buffers should be picked earlier
Adds a 0.5s advantage for audio buffers to being
picked earlier for muxing.
2009-11-24 12:44:56 -03:00
Robert Swain
98279be735 qtdemux: Fix push mode by making sure stbl information is available in
next_entry_size ()
2009-11-24 16:40:19 +01:00
Robert Swain
db5de8f1b6 qtdemux: Fix order of arguments in log message 2009-11-24 16:35:20 +01:00
Robert Swain
f9745e89d3 qtdemux: Ease debugging by removing a goto for an error message 2009-11-23 16:29:15 +01:00
Robert Swain
4025d7cbd7 qtdemux: Parse per sample rather than all at once but build complete index when
seeking
2009-11-23 16:29:15 +01:00
Robert Swain
0c62109d20 qtdemux: Save atom data for later use so it doesn't get freed after initial
parsing
2009-11-23 16:29:15 +01:00
Robert Swain
29c33806c1 qtdemux: Parse from the previously parsed sample up to sample n 2009-11-23 16:29:14 +01:00
Robert Swain
52b1040219 qtdemux: Make qtdemux_parse_samples () parse up to n samples 2009-11-23 16:29:14 +01:00
Robert Swain
1f7b878d89 qtdemux: Separate off stbl sub-atom initialisation 2009-11-23 16:29:14 +01:00
Robert Swain
6a6d2c4970 qtdemux: Move variables into context in preparation for refactorisation 2009-11-23 16:29:14 +01:00
Robert Swain
ab61fb22f6 qtdemux: Fix bug where stps is never parsed due to logic error 2009-11-23 16:29:14 +01:00
Robert Swain
a1e2047472 qtdemux: Port ctts from Gnode * to GstByteReader 2009-11-23 16:29:14 +01:00
Robert Swain
9e49197208 qtdemux: Switch from QtAtomParser to GstByteReader 2009-11-23 16:29:14 +01:00
Wim Taymans
5d41590601 qtdemux: fix typo and grammar 2009-11-23 12:53:50 +01:00
Tim-Philipp Müller
5908c40405 deinterlace: fix typo in mode enum description 2009-11-20 10:30:00 +00:00
Stefan Kost
9ee0815e85 docs: more links and better short description
Fix spelling of GstRtpSsrcDemux to get it linked. Add more links. Change
the short description to be more meaningful.
2009-11-20 11:25:49 +02:00
Thiago Santos
e35085e5b5 qtdemux: Add more fields to SVQ3 caps
qtdemux only added the whole stsd atom as 'codec_data'
in its output caps for SVQ3. This patch makes it add
the SEQH (inside a SMI atom) and a gamma field (taken
from the gama atom) if available.

Fixes #587922
2009-11-18 16:41:50 -03:00
Edward Hervey
f2f75d7fd9 wavenc: Raise rank of muxer to PRIMARY 2009-11-18 17:55:42 +01:00
Edward Hervey
8a1e0c53ae y4m: Raise rank of encoder to PRIMARY 2009-11-18 17:54:36 +01:00
Edward Hervey
a5dd867d6f law: Raise rank of encoders to PRIMARY 2009-11-18 17:54:35 +01:00
Bastien Nocera
efc611e420 Add user-id and user-pw properties
So that one doesn't need to modify the URL to have access
to authenticated RTSP streams.

fixes #601728
2009-11-18 17:27:19 +01:00
Mark Nauwelaerts
bf5f3a3964 qtdemux: fix bogus memory chunk size check 2009-11-18 12:54:48 +01:00
Wim Taymans
f52859432f jitterbuffer: release lock before emiting signals
Release the jbuf lock before emiting the request-pt-map signal to avoid
deadlocks. We also need to catch the shutdown case when locking again.

Fixes #593354
2009-11-18 10:50:44 +01:00
Wim Taymans
8c3b03de26 rtp: add BroadcomVoice depayloader 2009-11-18 10:50:43 +01:00
Wim Taymans
039d225a78 rtpbvpay: add rfc reference 2009-11-18 10:50:43 +01:00
Wim Taymans
02476fb5a3 rtp: add BroadcomVoice payloader 2009-11-18 10:50:43 +01:00
Jan Urbański
dd82612340 flvmux: properly finish the ECMA array
The ECMA array with the file index was missing a mandatory end marker.
Fixes bug #601242.
2009-11-18 08:03:43 +01:00
Jan Schmidt
baa79ffecb Use new still-frame API from gst-plugins-base 2009-11-18 03:09:06 +00:00
Michael Smith
fe9415544e qtdemux: identify IMA adpcm in qt properly. 2009-11-17 17:59:13 -08:00
Tim-Philipp Müller
4b1566d7f3 equalizer: printf format fix 2009-11-05 23:40:15 +00:00
Thiago Santos
feed8c2af3 avimux: do not write empty INFO list
avoid writing an empty INFO list chunk, both because
it is useless and because vlc refuses to play the
resulting file.
2009-11-05 12:31:56 -03:00
Sebastian Dröge
fb682d0444 equalizer: Notify about band property changes caused by changing number of bands 2009-11-05 10:54:12 +01:00
Sebastian Dröge
64e00f172c equalizer: Make changes to band properties and the number of bands threadsafe 2009-11-05 10:45:59 +01:00
Sebastian Dröge
025e26f73a equalizer: Fix stupid off by two bug 2009-11-05 10:30:46 +01:00
Sebastian Dröge
9405a328b1 equalizer: Add band property to select the band filter type
This allows per band configuration of a peak, low shelf or
high shelf filter, which can be very useful if the band frequencies
and widths are manually configured.
2009-11-05 08:21:33 +01:00
Sebastian Dröge
0525abd4af equalizer: Fix code style 2009-11-05 08:21:33 +01:00
Sebastian Dröge
e1acc8f4da equalizer: Some cleanup 2009-11-05 08:21:33 +01:00
Gabriel Millaire
773f142483 celtpay/depay : change GST_DEBUG_OBJECT to GST_LOG_OBJECT in pay_handle_buffer and depay_process 2009-11-04 12:02:50 -05:00
Gabriel Millaire
ac90398092 celtpay/depay: Negotiate parameters through caps
celtdepay : added default framesize(480) channels(1) and clockrate(32000)
            depay_setcaps : now gets channels and framesize from string with default value
            depay_process : now adds timestamp to outbuf
            Added frame_size to GstRtpCeltDepay
            Changed some GST_DEBUG to GST_DEBUG_OBJECT or GST_LOG_OBJECT
celtpay : getcaps : gets channel and framesize and sets caps
          Added frame-size to static caps for audio/x-celt
2009-11-04 12:02:50 -05:00
Jan Schmidt
1636bb0800 deinterlace: Pull in CFLAGS and LIBS flags from -base before core before system. 2009-11-04 15:59:49 +00:00
Edward Hervey
8df3e5c22b qtdemux: init variables to make compiler on osx build bot happy 2009-11-04 16:47:42 +01:00
Tim-Philipp Müller
261454dd92 qtdemux: init variables to make compiler on osx build bot happy 2009-11-03 16:05:47 +00:00
Tim-Philipp Müller
65a1db99eb deinterlace: remove pointless call to gst_element_no_more_pads() 2009-11-02 08:45:53 +00:00
Stefan Kost
03d2f4bdec level: fix decay to be smooth
The length not having any fractional part as it was promoted to gdouble after
dividing two guint64.
2009-11-01 00:31:48 +02:00
Stefan Kost
71044b37b6 level: calculate the message-intervall when it changes 2009-11-01 00:31:48 +02:00
Stefan Kost
f5b3392fa6 level: clocktime is a guint64, use right macro to init fields 2009-11-01 00:31:48 +02:00
Stefan Kost
519e424494 level: use more g-style types 2009-11-01 00:31:48 +02:00
Wim Taymans
0c12f585e3 avidemux: use segment_full when we can
Use segment_full so that we can pass the applied rate to the segment values. We
will change the applied rate when we implement skip mode.
2009-10-27 18:07:18 +01:00
Robert Swain
0cbe0d6e98 wavenc: Fix buffer offset by moving length incrementation 2009-10-27 12:43:33 +01:00
Michael Smith
b0b54d9324 Add dependencies of gstriff to things that link to gstriff, needed on Win32. 2009-10-23 18:09:43 -07:00
Stefan Kost
e43eb89449 tests: add a jitterbuffer test
Tests pushing a few buffers in various order and asserting the order sent by the
jitterbuffer. Contains two disabled tests that need more work.
2009-10-22 13:35:57 +03:00
Sebastian Dröge
68176befa2 matroskamux: Dirac "muxing" units end on EOS too
A Dirac muxing unit are all non-picture, non-end-of-sequence
packets up to and including the first picture or eos packet.

See http://www.diracvideo.org/wiki/index.php/ContainerFormatMappingGuidelines
2009-10-22 12:32:32 +02:00
Tim-Philipp Müller
457ac565ba avidemux: fix compilation with debugging disabled
total_idx is always evaluated.
2009-10-22 02:09:08 +01:00
Edward Hervey
683f2a02fb avidemux: Stop scanning at the last entry... and not the one before :)
This ensures we actually push out everything
2009-10-20 18:23:28 +02:00
Andy Wingo
c917d65e6d qtdemux: unpack more information into image/x-j2c caps
* gst/qtdemux/qtdemux_fourcc.h: Add new fourccs for use by the mj2
  unpacker.
* gst/qtdemux/qtdemux.c (qtdemux_parse_trak): Unpack JPEG2000 component
  mapping and channel definitions from the jp2h header. Will add
  component-map and channel-definitions elements to the caps if the
  component maps or channel definitions are nonstandard, where standard
  order means RGB, 444 packed YUV, or greyscale, with no alpha channel.

Fixes #598915.
2009-10-20 17:20:55 +02:00
Stefan Kost
217b54a8f6 level: code cleanup
Use gdouble instead of double. Calculate falloff_time once instead of twice.
2009-10-18 23:53:42 +03:00
Edward Hervey
024f1bae0c avidemux: MEMDUMP the junk blobs
It will only actually pull the junk blobs from upstream if the memdump
level is activated
2009-10-18 16:16:43 +02:00
Edward Hervey
1f5ace4de1 avidemux: Some avi files have INFO lists in the headers. 2009-10-18 16:16:43 +02:00
Edward Hervey
6e849f84fc avidemux: Don't seek on empty streams 2009-10-18 16:16:43 +02:00
Edward Hervey
a6ed612f42 avidemux: Ensure _calculate_durations_from_index only uses valid streams 2009-10-18 16:15:08 +02:00
Edward Hervey
1936d6ed26 avidemux: Only call convert function if we have strf.auds 2009-10-18 16:15:08 +02:00
Edward Hervey
af99a4a1de avidemux: Use first indexed stream for seeking.
In the future, main_stream can be adjusted to contain the optimal stream
as mentionned in the FIXME line 3440
2009-10-18 16:15:05 +02:00
Edward Hervey
2110cbe556 avidemux: Only expose streams that actually have something in it.
This guarantees that in pull-mode, all streams have a valid index to
work with.
2009-10-18 16:14:40 +02:00
Edward Hervey
546aa4c4dd avidemux: Properly mark presence of index.
Instead of blindly saying we have an index, only do so if we have a
non-empty index.
2009-10-18 15:40:37 +02:00
Mark Nauwelaerts
3d0659b813 debugutils: register pushfilesrc element 2009-10-16 18:19:20 +02:00
Mark Nauwelaerts
8f2beb5e51 avimux: support (some) VBR audio muxing
AVI format can handle VBR audio provided audio chunks are of fixed duration
(cfr fixed duration video frames).  Apply this approach to (always) parsed
raw AAC and (if parsed) to MPEG-1/2 audio.

See #368681.
2009-10-16 17:31:02 +02:00
Stefan Kost
6904e46ef2 build: use gst-glib-gen.mak to fix the glib build rules.
The build rules in glib-gen.mak were using pattern rules in a non save way.
2009-10-16 11:53:38 +03:00
Mark Nauwelaerts
7ceeb14834 avidemux: adjust flow return aggregation to updated loop_data
In particular, each stream is now treated separately, and one stream's
EOS should not lead to overall EOS.
2009-10-15 21:32:08 +02:00
Mark Nauwelaerts
354a062c89 qtdemux: check some more atom sizes prior to parsing 2009-10-15 17:06:41 +02:00
Wim Taymans
6725c91387 rtsp: handle events in TCP mode
We need to handle events in TCP mode so that we can reply to the LATENCY event
with TRUE.
2009-10-15 13:20:26 +02:00
Mark Nauwelaerts
f071ff6993 avidemux: add missing argument in debug message 2009-10-15 11:26:09 +02:00
Wim Taymans
88884cfddb rtspsrc: forward events into the rtpbin
Only catch the SEEK event on the srcpad and let other events enter the rtpbin.
2009-10-14 17:01:51 +02:00
Thiago Santos
959a3f9c95 matroskademux: Fix late tags finding
Use the correct taglist variable when notifying of late tags.
2009-10-14 11:33:24 -03:00
Mark Nauwelaerts
0141934eec avidemux: use GstIndex for (limited) seeking in push mode
... but disable this for now.  Although it basically works fine,
user experience might be shaky (depending on taste), since there
is no keyframe info in push mode.
2009-10-14 13:15:09 +02:00
Mark Nauwelaerts
35dc28d69a avidemux: add GstIndex support 2009-10-14 13:15:06 +02:00
Mark Nauwelaerts
92dd51e511 avidemux: also determine duration in push mode 2009-10-14 13:15:04 +02:00
Mark Nauwelaerts
e967767b27 qtdemux: add GstIndex support 2009-10-14 13:15:02 +02:00
Håvard Graff
58b9de4cca rtpptdemux: only forward the lost-event to the last seen pt-number
forward all events on all pads except for the PacketLost event, which we want to
forward to the last seen pt pad.

Fixes #598377
2009-10-14 12:28:55 +02:00
Wim Taymans
daa6d8f206 avidemux: demote some warnings to debug 2009-10-13 18:19:32 +02:00
Wim Taymans
9aa151a661 avi: add new avi flag we might want to use 2009-10-13 17:48:51 +02:00
Wim Taymans
df0335e65b avimux: calculate suggested buffer size
Calculate the suggested buffer size based on the largest chunk in the file.

See #597847
2009-10-13 17:48:51 +02:00
Wim Taymans
b134ca31fa avimux: add jpeg2000 to allowed caps 2009-10-13 17:48:51 +02:00
Wim Taymans
aea78a75ac avidemux: add debug for the superindex offsets 2009-10-13 17:48:50 +02:00
Jan Schmidt
99f43dbb58 qtdemux: Fix uninitialized variable warning
Fix another bogus may-be-used-uninitialized warning in qtdemux
2009-10-13 16:03:13 +01:00
Wim Taymans
50110d022d avi: lower max file size
Make a constant of the max file size and lower the value to what ffmpeg does,
hopefully improving compatibility with windows media player.

See #597847
2009-10-13 13:08:33 +02:00
Jan Schmidt
42b09362f6 qtdemux: Fix uninitialized variable warnings
The gcc on the OS/X buildbot complains about these variables not being
initialized, even though they can't possibly actually be used
uninitialized.
2009-10-13 00:12:42 +01:00
Mark Nauwelaerts
6f34e2b0db qtdemux: also consider Quicktime text subtitles 2009-10-09 17:49:20 +02:00
Mark Nauwelaerts
955a719c1a qtdemux: provide language tag for stream 2009-10-09 17:49:17 +02:00
Mark Nauwelaerts
1210a92ff6 qtdemux: refactor common parts in track parsing 2009-10-09 17:49:14 +02:00
Mark Nauwelaerts
5ed2c3e562 qtdemux: refactor buffer processing and sending
... so it can be used in both pull and push based mode.
2009-10-09 17:49:12 +02:00
Mark Nauwelaerts
674b0c4289 qtdemux: extract palette data for dvd subpicture streams
... and send it downstream using custom dvd event
2009-10-09 17:49:10 +02:00
Mark Nauwelaerts
b2d70862e8 qtdemux: support 3GPP timed text subtitles
In particular, also make subtitle support less subp(icture)-centric.
2009-10-09 17:49:06 +02:00
Mark Nauwelaerts
faaa32dccb qtdemux: NULL is not a valid taglist 2009-10-09 17:49:04 +02:00
Mark Nauwelaerts
533106203c qtdemux: recognize some more encypted track cases 2009-10-09 17:49:02 +02:00
Josep Torra
114dbba7ad id3: fixes warnings building on macosx
Another round on the formating of that debug line.
2009-10-09 15:59:25 +02:00
Stefan Kost
53cb3e2716 id3: cast pointer math results to glong 2009-10-09 14:44:02 +03:00
Stefan Kost
f854836f5c buikd: explicitely cast, to tell some compilers that this is not long int 2009-10-09 14:21:09 +03:00
Stefan Kost
f41d7e7bd5 build: don't cast, but use the right format specified instead
This correct some of the previous macos fixes.
2009-10-09 13:54:24 +03:00
Josep Torra
863233abf5 rtpvrawpay: fix warning on macosx 2009-10-09 12:01:10 +02:00
Josep Torra
a1fbe64317 rtph263pay: fix warning on macosx 2009-10-09 11:57:59 +02:00
Josep Torra
c3d3eb6c3b qtdemux: fix warnings building on macosx 2009-10-09 11:54:03 +02:00
Josep Torra
093546ba74 id3demux: fix printf warnings on macosx 2009-10-09 11:43:45 +02:00
Josep Torra
28ccc40bab avidemux: fix warning in macosx making the format portable 2009-10-09 11:43:44 +02:00
Josep Torra
00aa3421e0 audiofx: use G_GUINT64_FORMAT to fix warnings on OSX 2009-10-09 11:43:44 +02:00
René Stadler
c40cb18762 matroskademux: fix strstr() usage on possibly unterminated string 2009-10-08 23:31:07 +03:00
Jan Schmidt
cdb0b68e21 avi/wav: Fix some compiler warnings about incompatible pointers. 2009-10-08 10:20:09 +01:00
Jan Schmidt
db6af4bd57 multifile: Fix plugin description 2009-10-07 23:42:48 +01:00
Stefan Kost
e0cdd879b4 build: fprintf, sprintf, sscanf need stdio.h 2009-10-07 14:03:20 +03:00
Stefan Kost
27ea0b076a equalizer: use shelfing filters for first and last band
Refactor the filter setup. Add two new filters with shelf characteristics for
first and last band. Change gain calculation as recommended in the quoted
document (no qrt needed). Rename variables to match the formulas in the
document.
2009-10-07 00:35:27 +03:00
Stefan Kost
7b6e594b69 equalizer: fix filter history usage. Fixes #597397
The process functions where overwriting the history for each channel. Also pull
some static things out of the inner loop.
2009-10-05 23:04:39 +03:00
Wim Taymans
0040d01265 rtpbin: use locking around the sessions 2009-10-05 16:07:24 +02:00
Tim-Philipp Müller
45ff905771 qtdemux: make sure compatible brands buffer exists before dereferencing it 2009-10-05 11:46:08 +01:00
Robert Swain
c7b5df91a9 qtdemux: fix printf warnings on OSX
Cast variables passed to printf to avoid warnings about incorrect
formats (most likely caused by sizeof returning a size_t).

Fixes #597348.
2009-10-05 00:35:15 +01:00
Tim-Philipp Müller
4590daf202 qtdemux: remove internal genre table
No need to maintain our own genre table in qtdemux. The genres are
identical to the ID3 genres, so we can just use libgsttag's
gst_tag_id3_genre_get() to look them up.
2009-10-05 00:26:44 +01:00
Robert Swain
c45c304a7e Fix printf formats to avoid warnings in avidemux. Fixes #597214
https://bugzilla.gnome.org/show_bug.cgi?id=597214
2009-10-03 17:25:19 +02:00
Sebastian Dröge
650292706d matroskademux: Change one GST_WARNING to a GST_DEBUG 2009-10-03 12:21:34 +02:00
Sebastian Dröge
48b784e715 flvdemux: If there's no audio stream after 6 seconds of video signal no-more-pads
...and the other way around. Also ignore any audio/video streams that appear
after no-more-pads.

Fixes bug #597091.
2009-10-03 12:21:34 +02:00
Sebastian Dröge
f84bc538b5 flvdemux: Make sure to only signal no-more-pads a single time 2009-10-03 12:21:34 +02:00
Stefan Kost
d1d126b5b4 rtp: add missing include to fix the build 2009-10-02 18:25:16 +03:00
Stefan Kost
da05a85455 videofilter: add G_OBJECT_WARN_INVALID_PROPERTY_ID to property setter 2009-10-02 13:44:41 +03:00
Stefan Kost
948d5168ce level: don't give wrong number of fields in the message docs 2009-10-02 13:44:41 +03:00
Wim Taymans
8fb77403c5 jitterbuffer: cache latency in nanoseconds
Cache the latency in nanoseconds units to avoid having to convert the
milliseconds value to nanoseconds all the time.
2009-10-01 12:52:40 +02:00
Wim Taymans
c262735164 jitterbuffer: handle -1 input timestamps
Don't try to check a -1 timestamp against the max delay.
2009-10-01 12:12:09 +02:00
Stefan Kost
458cd4dcdc avi: don't misues perf-category and remove unused ext category
The performance category is meant to be used to audit codepaths that lead to bad
performance (e.g. copies, conversion that can be avoided).
Remove the event category which is not used.
2009-10-01 10:57:42 +03:00
Olivier Crête
00db9a585b rtpg729pay/depay: Demote per-buffer debug messages to log level 2009-09-30 20:36:05 -04:00
Olivier Crête
165516f0ef rtpg729pay: Don't leak incoming buffers after subbuffering them 2009-09-30 20:36:05 -04:00
Olivier Crête
680c97a7ca rtpg729pay/depay: Add debug categories 2009-09-30 20:36:05 -04:00
Olivier Crête
1ba7693f7a rtpg729pay: Remove long unneeded define replacement 2009-09-30 20:36:05 -04:00
Wim Taymans
3f263edbbf avi: small cleanups 2009-09-28 22:18:25 +02:00
Wim Taymans
217315c20b avi: fix timestamping in some audio streams
For vbr audio streams we need to use the number of blocks to calculate the
timestamps.
When the allocation of additional index memory fails, don't throw away what
we had before.
Various cleanups.
2009-09-28 22:17:02 +02:00
Wim Taymans
7b9b8343ba avi: add support for ODML indexes again 2009-09-28 22:17:00 +02:00
Wim Taymans
ceb7d66e25 avi: implement index scanning
Implement scanning of the file when we can parse the index.
Some refactoring of common code.
Cleanups and comments.
Remove some reimplemented code.
Remove index massage code and put a FIXME where we should do something
equivalent later.
2009-09-28 22:16:57 +02:00
Wim Taymans
8aa3830852 avi: fix reverse playback 2009-09-28 22:16:55 +02:00
Wim Taymans
3338f91cfe avi: fix prev keyframe search and cleanups 2009-09-28 22:16:53 +02:00
Wim Taymans
1b325945e5 avi: remove code that got converted 2009-09-28 22:16:50 +02:00
Wim Taymans
c199b1d039 avi: more cleanups
Remove some duplicate counters.
Be smarter when updateing the current the timestamp and offset in the stream
because we can reuse previously calculated values when simply go forward one
step.
Correctly set metadata on outgoing buffers.
2009-09-28 22:16:48 +02:00
Wim Taymans
0d70fe30a8 avidemux: small cleanups 2009-09-28 22:16:46 +02:00
Wim Taymans
b4a490655a avi: fix read offset and cleanups 2009-09-28 22:16:43 +02:00
Wim Taymans
9c37611dfa avi: rewrite index playback
disable code, start on reimplementing loop based operation.
Rewrite the index handling so that all streams use their own index for decoding
media.
2009-09-28 22:16:41 +02:00
Wim Taymans
89bcbbbe7c avidemux: add new index parsing code
Add a new function and datastructure to parse and hold the index entries on a
per stream base. Also avoid doing too much work trying to figure out the
timestamps and durations as we can trivially do that later.

Less information in the entries makes them 2 times smaller and not doing too
much work makes this code about 12 times faster than the regular case.

Hook in the new function alongside the existing function for comparison until
the rest of the code is updated to handle the new index datastructure.
2009-09-28 22:16:38 +02:00
Mark Nauwelaerts
0fac7b5347 qtdemux: some optional QT specified stsd MPEG-4 atoms also apply to H264
Fixes #596319.
2009-09-25 19:23:15 +02:00
Mark Nauwelaerts
e21d16a4f8 qtdemux: only send tag events downstream after newsegment 2009-09-25 16:47:42 +02:00
Mark Nauwelaerts
50d5c8dce5 rtspsrc: if transport protocol unsupported, try another one
Also change error message to more accurately reflect cases in which
it can occur.
2009-09-25 16:47:39 +02:00
Wim Taymans
03f46a42e5 qtdemux: add durations modulo 1<<32
For calculating the durations of each sample, we are supposed to add each
duration modulo 1<<32 so make the elapsed time counter a uint32.

Fixes #595942
2009-09-25 11:54:06 +02:00
Wim Taymans
4e114a2b24 qtdemux: small cleanup 2009-09-24 20:38:54 +02:00
Tim-Philipp Müller
01e00ba1cd qtdemux: don't use core API that doesn't exist yet
There's no gst_byte_reader_has_remaining() yet. Fixes build.
2009-09-24 19:33:39 +01:00
Tim-Philipp Müller
fab4113c24 qtdemux: map some atomparser functions to their new bytereader equivalents
Now that GstByteReader has unchecked and inlined variants as well, map
atomparser functions to their respective bytereader equivalents.
2009-09-24 16:34:08 +01:00
Tim-Philipp Müller
0f197776e1 qtdemux: add qt_atom_parser_has_chunks() and fix indentation 2009-09-24 16:32:02 +01:00
Tim-Philipp Müller
f65e6ea3a1 qtdemux: bail out instead of trying to alloc silly index sizes
If it looks like we would be allocating a silly size for our sample
index, just bail out instead of trying to allocate it. Helps with
broken or fuzzed files where we might end up trying to malloc a
couple of hundred MBs otherwise.
2009-09-24 16:29:26 +01:00
Tim-Philipp Müller
abaf91e428 qtdemux: error out correctly if we don't even have enough bytes for an atom header 2009-09-24 16:29:25 +01:00
Tim-Philipp Müller
25db7df49b qtdemux: init fourcc to 0 as well to avoid invalid reads when printf'ing error message 2009-09-24 16:29:25 +01:00
Tim-Philipp Müller
9da3ed6491 qtdemux: add qt_atom_parse_has_remaining() to avoid overflows with _get_remaining() 2009-09-24 16:28:40 +01:00
Tim-Philipp Müller
a16feec38e qtdemux: use GstByteReader when parsing tkhd atom 2009-09-23 16:54:43 +01:00
Tim-Philipp Müller
6b7f4f5e23 qtdemux: use unsigned ints for node length and do more sanity checking of the atom length 2009-09-23 16:54:43 +01:00
Tim-Philipp Müller
3abeb1e578 qtdemux: use GstByteReader for atom dumping and fix a few bugs 2009-09-23 16:54:42 +01:00
Tim-Philipp Müller
c8c9b0f35d qtdemux: move stco, stts, stss and stps atom parsing over to GstByteReader
Make sure we don't read beyond the atom boundary. Note that the code
behaves slightly differently in the corner case where there is not
enough atom data for the specified number of samples (n_samples_time)
in the atom, but still enough data to fill the pre-allocated index of
n_samples entries: before we would just stop parsing the stts data
and continue, whereas now we will likely error out. This should not
be a problem in practice though. We could maintain the old behaviour
by doing reads with a size check inside the loop if needed.
2009-09-23 16:54:42 +01:00
Tim-Philipp Müller
4be46b1586 qtdemux: use bytereader to parse stsz and stsc atoms
Use GstByteReader to parse stsz and stsc chunks, and check size of
available data before parsing it, instead of blindly assuming there
will be enough data. Fixes crashes with some fuzzed/broken files.
2009-09-23 16:54:42 +01:00
Tim-Philipp Müller
5875e2016a qtdemux: add qt_atom_parser_get_offset() and optimise _peek_sub() 2009-09-23 16:54:42 +01:00
Tim-Philipp Müller
410ebb7eb3 qtdemux: add QtAtomParser, an inlined GstByteReader variant 2009-09-23 16:54:41 +01:00
Mark Nauwelaerts
02581dd2a5 matroskademux: use proper order for no-more-pads and newsegment and tag sending 2009-09-23 17:24:22 +02:00
Mark Nauwelaerts
702df566c3 matroskademux: sprinkle a few branch prediction macros 2009-09-23 17:24:22 +02:00
Alessandro Decina
195883b30a Fix compile warnings with gcc 4.0.1. 2009-09-22 15:04:36 +02:00
Jan Schmidt
600516be90 matroskamux: Don't get stuck in an infinite loop with Dirac
At the end, Dirac streams have an EOS packet with 0 length.
Don't ever sit in an infinite loop when processing one. Allows
muxing Dirac into mkv to complete successfully.
2009-09-22 11:50:11 +01:00
Tim-Philipp Müller
0506545b04 videomixer: fix up Makefile some more
Remove CFLAGS from LIBADD and make order of the various CFLAGS and
LIBS at least consistent with each other.
2009-09-22 11:02:02 +01:00
Brian Cameron
341be447a6 videomixer: Add $(GST_PLUGINS_BASE_LIBS) to LDFLAGS for linking libgstvideo
Fixes bug #595897.
2009-09-22 08:09:39 +02:00
Wim Taymans
10eb1a0ff4 avi: fix timestamps in push mode 2009-09-21 18:10:12 +02:00
Wim Taymans
2f26ee4285 avi: add some performance measurements
Measure the performance of various index and header parsing steps to the
PERFORMANCE debug category.
2009-09-21 12:32:51 +02:00
Stefan Kost
0868ddf30f avidemux: some logging cleanup to help understanding the index parsing overhead 2009-09-18 14:27:45 +03:00
Olivier Crête
750387f520 rtpg729pay: Fix adapter leak
The adapter would be leaked if it was empty and the data could be pushed out directly.
2009-09-15 17:24:24 -04:00
David Schleef
78eeb6636e multifilesink: Add next-file property
Add a property to allow control over what event causes a file
to finish being written and a new file start.  The default is
the same as before -- each buffer causes a new file to be
written.  Added is a case where buffers are written to the
same file until a discontinuity in the stream.
2009-09-13 20:00:53 -07:00
Michael Smith
3257374310 wavparse: treat a zero-sized data chunk as extending to the end of the file.
This fixes playback of some files that don't have a valid data chunk length,
apparently some program creates these.
2009-09-11 13:34:01 -07:00
Wim Taymans
445236a769 spectrum: add post-messages property
Add a post-messages property and deprecate the less descriptive message
property.
2009-09-11 13:28:35 +02:00
Wim Taymans
1935483fbf multifilesink: rename silent to post-messages
Use the post-messages property name instead of silent as it is more
descriptive.
2009-09-11 13:12:54 +02:00
Wim Taymans
f68cd7e708 multifilesink: post messages for each buffer
Add a silent property that can be set to FALSE to post messages on the bus for
each written file.
Do some more cleanups.
Add some docs.

Fixes #594663
2009-09-11 12:17:21 +02:00
Olivier Crête
411c71da13 rtph263pay: Allocate Boundry structs on the stack instead of the heap to avoid leaks
Fixes bug #594691.
2009-09-11 07:31:38 +02:00
Stefan Kost
0a7ef67ad0 docs: fix gtk-doc warnings 2009-09-10 10:28:48 +03:00
Sebastian Dröge
a9909c1abf videobox: Fix AYUV->I420 conversion
For this fix the averaging of the chroma values. It should't be (a/2 + b)/2
but just (a + b)/2.

Fixes bug #594599.
2009-09-09 16:28:53 +02:00
Marc-André Lureau
fe2d8bdc64 multipartmux: mark data buffer as delta-unit
So that multifdsink always start sending header buffer first

Fixes #594520
2009-09-08 18:34:49 +02:00
Marc Leeman
6b46aeb6a3 rtpbin: add ignore-pt parameter
Add a parameter 'ignore-pt' that disables creating a gstrtpptdemux module and
ghosts the pads of gstrtpjitterbuffer instead of the ones of gstrtpptdemux.

Fixes #594490
2009-09-08 17:38:32 +02:00
Håvard Graff
2912b21d14 rtpbin: propagate payload-type-change signal from demuxer
fixes #594254
2009-09-08 13:59:56 +02:00
Havard Graff
a52309eff7 jitterbuffer: change severity of clock-rate change debug
Make log GST_DEBUG under normal circumstances, GST_WARNING otherwise.

Fixes #594253
2009-09-08 13:44:49 +02:00
Håvard Graff
40549278c3 jitterbuffer: avoid throwing reordered buffers with same timestamps
When we receive a reordered packet with the same timestamp as the previous one
(which can happen for fragmented packets) don't consider the packet as lost but
instead wait for the reordered packet to arrive.

Switch the warning-level, so that a reordering does not get a warning, only
an actual produced lost-packet.

Fixes #594251
2009-09-08 13:39:31 +02:00
Havard Graff
6108024838 rtpjpegdepay: add missing math.h include
Fixes #594247
2009-09-08 13:32:51 +02:00
Arnout Vandecappelle
19455200b1 rtspsrc: fix memory leak
In gst_rtspsrc_parse_digest_challenge(), rtspsrc does a g_strndup of the auth
header items and then passes them to gst_rtsp_connection_set_auth_param()
without freeing.

Fixes #594133
2009-09-08 13:30:29 +02:00
Stig Sandnes
8f3299c547 rtpbin: make free_session() remove stream references
When receiving a sync-packet, all sessions with the same cname will be compared
and synced together. In this process, there could still be references to a
session that has been shut down in the meanwhile.

This patch makes sure that these references are removed when shutting down a
session, so that the syncing can be done safely.

Fixes #594283
2009-09-08 13:18:29 +02:00
Havard Graff
e08e610db0 rtpbin: use locked state on internal bins
Set the locked state on internal elements to make sure that they don't change
back to another state when shutting down.

Fixes #594248
2009-09-08 12:41:52 +02:00
Zaheer Merali
c6b2dff77e y4menc: Add interlaced support
Fixes #591713

Signed-off-by: David Schleef <ds@schleef.org>
2009-09-05 20:53:10 -07:00
David Schleef
55d2754098 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.
2009-09-05 20:53:10 -07:00
Mark Nauwelaerts
868a4b1303 qtdemux: prevent a spurious debug warning 2009-09-04 13:51:25 +02:00
Sebastian Dröge
b35b752c41 matroskademux: Correctly handle NULL GstIndex 2009-09-04 07:10:03 +02:00
Laurent Glayal
371875c57a rtpsource: fix memleak
Don't leak the input buffer when the received and expected seqnum are different when
in probation.

fixes #594039
2009-09-03 19:37:10 +02:00
Olivier Crête
f542f710cf rtpjitterbuffer: Lock clock_rate variable
The priv->clock_rate variable could become -1 between when its checked to not
be -1 and when its used, causing an assertion. Fixed by taking the mutex
earlier in the chain() function.

Fixes #593955
2009-09-03 19:17:00 +02:00
Wim Taymans
3fcde4486d rtpsource: whitespace fixes 2009-09-03 19:17:00 +02:00
Wim Taymans
bf73a6ee3a rtpmpapay: whitespace fixes 2009-09-03 19:17:00 +02:00
Wim Taymans
3f629f6001 rtpsession: whitespace fixes 2009-09-03 19:16:59 +02:00
Stefan Kost
272683ff36 flvmux: fully use tagsetter to manage the tags. Fixes #563221
There is no need to manage a separate taglist.
2009-09-03 14:48:14 +03:00
Peter Kjellerstedt
fdf18653b7 rtpmanager: Fixed a copy & paste error 2009-09-01 15:06:46 +02:00
Peter Kjellerstedt
dc4f9575be rtpmanager: Removed unused variable priv
The variable priv was initialized in a lot of functions but then never
used for anything.
2009-09-01 13:21:23 +02:00
Peter Kjellerstedt
57adc2a803 rtpmanager: A little clean up
Make the code flow of gst_rtp_session_send_rtcp() and
gst_rtp_session_sync_rtcp() identical.
2009-09-01 13:04:14 +02:00
Peter Kjellerstedt
923b5b495a rtpmanager: Make sure that used caps are not freed already (take 2)
This reintroduces the fix for bug #593391. It also applies it in
gst_rtp_session_sync_rtcp() which has very similar code to
gst_rtp_session_send_rtcp().
2009-09-01 13:04:14 +02:00
Wim Taymans
8d924611e7 jitterbuffer: make sure time does not go backwards
When we construct a timestamp that would result in a timestamp that is earlier
than when the packet was received, reset the skew calculation as this is
probably a sign that the sender restarted or paused.

Fixes #593354
2009-09-01 12:48:28 +02:00
Peter Kjellerstedt
bfb1260af4 rtpmanager: Set caps in gst_rtp_session_send_rtcp() correctly again
The test for when to set an RTCP caps on the output pad in
gst_rtp_session_send_rtcp() accidentally got inverted in the last commit.
2009-09-01 11:32:41 +02:00
Sebastian Dröge
e7efa0a5be qtdemux: Add support for QCELP audio
Fixes bug #593757.
2009-09-01 10:26:46 +02:00
Peter Kjellerstedt
fbefd9c666 effectv: Fix compilation with gcc 3
Recent changes in gst-plugins-good/gst/effectv prevents it from being compiled
with gcc 3. The problem is that the new code uses preprocessor conditionals
within a macro call which does not work with older versions of gcc.

Fixes bug #593688.
2009-08-31 18:11:28 +02:00
Mark Nauwelaerts
c9a434bbff rtpmp4gdepay: consider (optional) auxiliary data when parsing 2009-08-31 16:50:01 +02:00
Mark Nauwelaerts
30efa405f3 rtpmp4gdepay: handle broken AU-Index in non-interleaved streams
In case of non-interleaved (= sequentially payloaded) streams,
the AU-Index serves little purpose (that is not already covered by
RTP fields).  (Broken) Payloaders might consider this field then
to be disregarded and have non spec compliant values, e.g. each
RTP packet having AU-Index 2 (rather than 0).  As such, ensure/force
simple sequential sending of non-interleaved streams.
2009-08-31 16:50:01 +02:00
Mark Nauwelaerts
15fa7d33ed qtdemux: also extract ftyp info in push mode 2009-08-31 16:50:01 +02:00
Mark Nauwelaerts
c469f6b38d qtdemux: consider 3gpp style tag parsing in some more cases
3GPP specs define a number of tags along with precise layout. While these
are normally expected to be found in a container whose major brand is a
3GPP brand, this may also happen when a 3GPP brand is only mentioned as a
compatible brand.  Apply some checks, heuristic and fallbacks to extract
such tags as well.
2009-08-31 16:50:00 +02:00
Mark Nauwelaerts
0f900afe1f wavparse: reflow exit, and fix some leaks 2009-08-31 16:50:00 +02:00
Mark Nauwelaerts
efb5d1b545 wavparse: push mode; add pad if needed so downstream gets EOS 2009-08-31 16:50:00 +02:00
Mark Nauwelaerts
79f69bbf72 wavparse: push mode; fix/improve chunk handling
Handle large, invalid or otherwise unusual chunk sizes.
Verify some chunk sizes to be at least the size they are
expected to be and round up some sizes to even number for
e.g. offset administration, which must also be properly
tracked in push mode.
2009-08-31 16:50:00 +02:00
Mark Nauwelaerts
bb2b02c5b7 avidemux: push mode; cater for unusual chunk sizes 2009-08-31 16:50:00 +02:00
Wim Taymans
a74c385b7b rtpsession: use proper locking for pads and caps
Use the sesion lock and shotdown variable to protect and ref the pads we are
going to push on.

fixes #561825
2009-08-31 16:38:27 +02:00
Wim Taymans
a522a2d4d2 rtpbin: whitespace fixes 2009-08-31 16:33:26 +02:00
Tim-Philipp Müller
4cf513da9b wavparse: clean up adapter properly
Reflow code so we don't try to clear or re-use an already-freed adapter.
2009-08-31 13:40:14 +01:00
Tim-Philipp Müller
d875e72b02 flactag, wavparse: GstAdapter is not a GstObject 2009-08-31 13:07:53 +01:00
Jan Schmidt
3f69f8d3ee flvdemux: Fix tests warning from setting a NULL index
Setting a null index in the tests was causing warnings by unreffing
NULL pointers. This is a bug exposed by a recent change in core, it
seems.
2009-08-31 12:10:05 +01:00
Wim Taymans
a26a2a9ff5 jitterbuffer: add slope estimation code and debug
Add some code to measure the sender speed vs the receiver speed. This can be
used to detect bursts.
2009-08-31 13:02:16 +02:00
Wim Taymans
4814d899c2 jitterbuffer: reset skew when timestamps change
Refactor the jitterbuffer resync code.
Reset the skew correction when we detect a big timestamp discont.

See #593354
2009-08-31 12:57:32 +02:00
Wim Taymans
e254936e34 jitterbuffer: make sure time never goes invalid
Since the skew can be negative, we might end up with invalid timestamps. Check
for negative results and clamp to 0.

See #593354
2009-08-31 12:47:15 +02:00
Jarkko Palviainen
1f14f577d8 udpsink: Add ttl multicast property
Add a new ttl-mc property to control the TTL on multicast addresses.

Fixes #588245
2009-08-31 12:16:01 +02:00
Jarkko Palviainen
e2518fedbe udp: split out TTL and loop options
Split setting the TTL and loop parameters in 2 methods as they are not related.
Fix setting the TTL correctly for multicast streams.

See #588245
2009-08-31 12:13:07 +02:00
Wim Taymans
6a53d0a2c9 rtp: whitespace fixes 2009-08-31 11:32:06 +02:00
Sebastian Dröge
867b8c9d15 videobox: Split declarations into a header file and add autocrop stuff to the docs 2009-08-31 08:19:25 +02:00
Sebastian Dröge
6976f3d39a videobox: Reconfigure basetransform if something changes again
For this invent a new lock and don't abuse the basetransform lock,
otherwise we'll end up in deadlocks.
2009-08-31 08:19:25 +02:00
Stephen Jungels
041ddd6f8f videobox: Add support for autocropping according to the caps
Fixes bug #582238.
2009-08-31 08:19:25 +02:00
Sebastian Dröge
041fa82179 rtpsession: Make sure that used caps are not freed already
Fixes bug #593391.
2009-08-31 08:09:09 +02:00
Sebastian Dröge
000a483d31 rtp: Use new gst_iterator_new_single() for the internal linked pads iteration 2009-08-31 08:09:09 +02:00
Sebastian Dröge
a1cddb3fd6 rtpsession: Use iterate internal links instead of deprecated get internal links 2009-08-31 08:09:09 +02:00
Sebastian Dröge
c8c02d2c7a jitterbuffer: Use iterate internal links instead of deprecated get internal links 2009-08-31 08:09:08 +02:00
Sebastian Dröge
97cb7bdb6c rtpssrcdemux: Use iterate internal links instead of deprecated get internal links 2009-08-31 08:09:08 +02:00
Wim Taymans
e9e94a771b qtdemux: add support for agsm
Fixes #592530
2009-08-21 11:44:43 +02:00
Mark Nauwelaerts
15d17763c0 qtdemux: fix qt style string tag extraction
QT style tags are tested on starting with (C) symbol using >>,
and (unsigned) int (may) have different >> behaviour.
Fixes #592232.
2009-08-18 19:01:11 +02:00
Olivier Crête
7f569ca9c8 rtpbin: Fix reference leak
Fixes #591476.
2009-08-14 13:47:18 +01:00
ric
92abe07e80 rtpsource: avoid buffer leak on bad seqnum
Fixes #590797
2009-08-11 02:30:47 +01:00
Wim Taymans
9f68303a2e rtpsource: allow for NULL caps on buffers
Add the NULL caps check where it matters and also cover another case of
potential NULL caps.

Fixes #590030
2009-08-11 02:30:47 +01:00
Olivier Crête
e37844fdc7 rtpsource: Incoming buffers do not always have caps 2009-08-11 02:30:47 +01:00
Wim Taymans
3091137217 rtpsession: avoid doing lip-sync in BYE
When we get a BYE packet, don't do lip-sync with the SR inside because some
senders have trouble constructing valid SR packets after BYE.
2009-08-11 02:30:47 +01:00
Wim Taymans
3747ede14a rtpbin: don't do lip-sync after a BYE
After a BYE packet from a source, stop forwarding the SR packets for lip-sync
to rtpbin. Some senders don't update their SR packets correctly after sending a
BYE and then we break lip-sync. We prefer to let the jitterbuffers drain with
the current lip-sync instead.
2009-08-11 02:30:47 +01:00
Wim Taymans
d2ef095b80 rtpbin: only reconsider once for BYE
When iterating the sources of a BYE packet, don't signal a reconsideration for
each of them but signal after we handled all sources.
2009-08-11 02:30:47 +01:00
Olivier Crête
e8c6bcdf8d rtpsession: Free conflicting addresses on finalize 2009-08-11 02:30:46 +01:00
Wim Taymans
428368b44a rtpbin: use new method for netaddress to string 2009-08-11 02:30:46 +01:00
Wim Taymans
512ba93159 rtpbin: do better cleanup of the src ghostpads
Connect to the pad-removed signal of the ptdemux elements so that we remove the
ghostpads for them. Fixes cleanup when going to NULL as well as when releasing
the sinkpads.

Fixes #561752
2009-08-11 02:30:46 +01:00
Wim Taymans
d7a8663e05 rtpsession: add a comment 2009-08-11 02:30:46 +01:00
Wim Taymans
c53e595d23 rtpbin: add SDES property
Remove all individual SDES properties and use one sdes property that takes a
GstStructure instead. This will allow us to add more custom stuff to the SDES
messages later.
2009-08-11 02:30:46 +01:00
Wim Taymans
9f330992f5 rtpbin: add SDES property that takes GstStructure
Remove all individual SDES properties and use one sdes property that takes a
GstStructure instead. This will allow us to add more custom stuff to the SDES
messages later.
2009-08-11 02:30:46 +01:00
Wim Taymans
d8496fb105 rtpbin: removed old gstrtpclient 2009-08-11 02:30:45 +01:00
Branko Subasic
779f67adc4 rtpbin: add support for buffer-list
Add support for sending buffer-lists.
Add unit test for testing that the buffer-list passed through rtpbin.

fixes #585839
2009-08-11 02:30:45 +01:00
Tim-Philipp Müller
c5793a6a45 Make build without warnings with debugging disabled 2009-08-11 02:30:45 +01:00
Olivier Crête
cf873498d2 rtpbin: Transform the right session sdes message
Fixes #584165
2009-08-11 02:30:45 +01:00
Olivier Crête
dee142a945 Add ssrc to application/x-rtp-source-sdes structure 2009-08-11 02:30:45 +01:00
Wim Taymans
bf15048f42 rtpsouce: the network address is in network order
Bring the network address in netowkr byte order to the host order.
2009-08-11 02:30:45 +01:00
Wim Taymans
91eef69131 rtpsource: byteswap the port from GstNetAddress
Since the port in GstNetAddress is in network order we might need to byteswap it
before adding it to the source statistics.
2009-08-11 02:30:45 +01:00
Wim Taymans
51251d0fa8 rtpbin: remove ptdemux ghostpads 2009-08-11 02:30:44 +01:00
Wim Taymans
7d9c2d20df rtpbin: add to new signal to remove SSRC pads 2009-08-11 02:30:44 +01:00
Ali Sabil
6c684e59c6 ssrcdemux: emit signal when pads are removed
Add action signal to clear an SSRC in the ssrc demuxer.
Add signal to notify of removed ssrc.

See #554839
2009-08-11 02:30:44 +01:00
Wim Taymans
48872d8215 rtpbin: use our ghostpads instead of its target
Since we keep a reference to our ghostpads, we can use them to track sessions.
This avoid us having to mess with the target of the ghostpad.
2009-08-11 02:30:44 +01:00
Wim Taymans
901b7f3b69 rtpbin: don't warn when getting request pads twice
Allow getting the request pads multiple times, just return the previously
created pads.
2009-08-11 02:30:44 +01:00
Wim Taymans
0ae6e3603b rtpsource: add RTP and RTCP source address
Add the RTP and RTCP sender addresses in the stats structure.
2009-08-11 02:30:44 +01:00
Wim Taymans
62727e8fab rtpsession: reuse source code for SDES
Reuse the RTPSource object property instead of duplicating code.
2009-08-11 02:30:44 +01:00
Wim Taymans
1719af9113 rtpbin: set target state on new elements
Set the state on newly added elements to the state of the parent.
Add some debug info and do some cleanups
2009-08-11 02:30:43 +01:00
Wim Taymans
9c92ee6209 rtpbin: unref requests pads after releasing 2009-08-11 02:30:43 +01:00
Olivier Crête
a1c0bb2488 rtpbin: Implement releasing the streams
See #561752
2009-08-11 02:30:43 +01:00
Olivier Crête
e77542d350 rtpbin: Keep jb signals handler
Keep the signal handlers so they can be disconnected at release time

See #561752
2009-08-11 02:30:43 +01:00
Wim Taymans
59d0590cd7 rtpbin: use the right lock for the sessions
Use the right lock when iterating the sessions.
2009-08-11 02:30:42 +01:00
Olivier Crête
a9d6f3558c rtpbin: Free session if request pads are released
Free the session when all the request pads are released.
Don't mess with the session list in free_session as it is called from a foreach
on that list.
Set the state of the upstream element to NULL first.

See #561752
2009-08-11 02:30:42 +01:00
Olivier Crête
46388b767f rtpbin: Implement relasing of the rtp recv pad 2009-08-11 02:30:42 +01:00
Olivier Crête
3509098468 rtpbin: Implement releasing of rtp send pads 2009-08-11 02:30:42 +01:00
Olivier Crête
2f6e9d7bf2 rtpbin: Implement release of the recv rtcp pad
See #561752
2009-08-11 02:30:42 +01:00
Olivier Crête
47d4bb90c1 rtpbin: Implement releasing of rtcp src pad
See #561752
2009-08-11 02:30:41 +01:00
Wim Taymans
11607c4d63 rtpssrcdemux: drop unexpected RTCP packets
We usually only get SR packets in our chain function but if an invalid packet
contains the SR packet after the RR packet, we must not fail but simply ignore
the malformed packet.

Fixes #581375
2009-08-11 02:30:41 +01:00
Olivier Crete
3482b47666 rtpsouce: make WARNING into LOG
Since neither rtpmanager nor any of the payloaders properly implement
pad allocation, there is no way for the rtpmanager to inform downstream elements
of the new SSRC if there is an SSRC collision. So the warning is emitted all the
time and it is confusing.

Fixes #580144
2009-08-11 02:30:41 +01:00
Olivier Crete
63636b1290 rtpsession: notify when SSRC changes
Emit a g_object_notify when the SSRc changes because of a collision.
Fixes #580144
2009-08-11 02:30:41 +01:00
Wim Taymans
d45d18c735 rtpsession: join the RTCP thread
Avoid a case where a joinable thread would be left unjoined, which leaked the
thread structure.
Fixes #577318.
2009-08-11 02:30:41 +01:00
Wim Taymans
64046416cc jitterbuffer: prevent overflow in EOS estimation
Use a guint64 instead of a guint to hold a 64bit value to prevent completely
bogues EOS estimation values due to overflows.
2009-08-11 02:30:41 +01:00
Wim Taymans
d6c623e90c rtpbin: we should not provide a clock
There is no need to provide a clock.
2009-08-11 02:30:41 +01:00
Wim Taymans
5ece6ae4e3 jitterbuffer: more estimated EOS fixes
Do more accurate EOS estimate and guard against backward timestamps.
2009-08-11 02:30:41 +01:00
Wim Taymans
cbad89600c jitterbuffer: release lock before pushing EOS
Make sure we release the jitterbuffer lock before we start pushing out data
because else we might deadlock.
2009-08-11 02:30:41 +01:00
Wim Taymans
918c9448f2 rtpbin: add on_npt_stop signal
Add the on_npt_stop signal to rtpbin and rtpjitterbuffer to notify the
application that the NPT stop position has been reached.
2009-08-11 02:30:41 +01:00
Wim Taymans
55c3da71c1 rtpbin: don't return FALSE on seek events
Silently ignore the seek event instead of returning FALSE.
2009-08-11 02:30:41 +01:00
Olivier Crête
109874ed50 gstrtpbin: Don't forward revc events to sender
Don't send events from the receiver to the sender side.
Fixes #572900.
2009-08-11 02:30:40 +01:00
Stefan Kost
7ae3923ac6 docs: various doc fixes
No short-desc as we have them in the element details.
Also keep things (Makefile.am and sections.txt) sorted.
Reword ambigous returns. No text after since please.
2009-08-11 02:30:40 +01:00
Wim Taymans
2c6ab34114 Send BYE packets immediatly for small sessions
When the number of participants is less than 50, the RFC allows for sending the
BYE packet immediatly instead of using the regular BYE timeout.
Fixes #567828.
2009-08-11 02:30:40 +01:00
Wim Taymans
7f0b100db5 Unlock the jitterbuffer before pushing out the packet-lost events.
Move some code before we do the unlock to make the jitterbuffer state
consistent while we are unlocked.
2009-08-11 02:30:40 +01:00
Olivier Crete
dfdc9b6662 gst/rtpmanager/: When an SSRC is found on the caps of the sender RTP, use this as the internal SSRC. Fixes #565910.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_setcaps_send_rtp), (create_send_rtp_sink):
* gst/rtpmanager/rtpsession.c: (rtp_session_set_internal_ssrc):
When an SSRC is found on the caps of the sender RTP, use this as the
internal SSRC. Fixes #565910.
2009-08-11 02:30:40 +01:00
Wim Taymans
0ad92e7da6 gst/rtpmanager/: Rename a method to better reflect what it really does.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_getcaps_send_rtp):
* gst/rtpmanager/rtpsession.c: (check_collision),
(rtp_session_schedule_bye_locked), (rtp_session_schedule_bye):
* gst/rtpmanager/rtpsession.h:
Rename a method to better reflect what it really does.
2009-08-11 02:30:40 +01:00
Wim Taymans
06d1532024 gst/rtpmanager/gstrtpsession.c: Use method to get the internal SSRC.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_getcaps_send_rtp):
Use method to get the internal SSRC.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_set_property), (rtp_session_get_property):
Add property to congiure the internal SSRC of the session.
Fixes #565910.
2009-08-11 02:30:40 +01:00
Wim Taymans
1786eb1e25 gst/rtpmanager/rtpsession.c: Only change the SSRC of the session and reset the internal source when the SSRC actually...
Original commit message from CVS:
* gst/rtpmanager/rtpsession.c: (rtp_session_set_internal_ssrc):
Only change the SSRC of the session and reset the internal source when
the SSRC actually changed. See #565910.
2009-08-11 02:30:40 +01:00
Wim Taymans
3fe87f7eab gst/rtpmanager/rtpsource.*: When no payload was specified on the caps but there was a clock-rate, assume the clock-ra...
Original commit message from CVS:
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(rtp_source_update_caps), (get_clock_rate):
* gst/rtpmanager/rtpsource.h:
When no payload was specified on the caps but there was a clock-rate,
assume the clock-rate corresponds to the first payload type found in the
RTP packets. Fixes #565509.
2009-08-11 02:30:40 +01:00
Arnout Vandecappelle
2142edd399 gst/rtpmanager/rtpjitterbuffer.*: Keep track of the last outgoing timestamp and of the last sender-side time. Timest...
Original commit message from CVS:
Patch by: Arnout Vandecappelle <arnout at mind dot be>
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew):
* gst/rtpmanager/rtpjitterbuffer.h:
Keep track of the last outgoing timestamp and of the last sender-side
time.  Timestamps can only go forward if they do at the sender
side, can only go back if they do at the sender side, and remain the
same if they remain the same at the sender side. Fixes #565319.
2009-08-11 02:30:40 +01:00
Wim Taymans
5b6700a022 gst/rtpmanager/rtpsession.c: Make obtain_source return an aditional ref so that we don't lose our ref to it when a se...
Original commit message from CVS:
* gst/rtpmanager/rtpsession.c: (obtain_source),
(rtp_session_create_source), (rtp_session_process_rtp),
(rtp_session_process_sr), (rtp_session_process_rr),
(rtp_session_process_sdes), (rtp_session_process_bye):
Make obtain_source return an aditional ref so that we don't lose our ref
to it when a session cleanup occurs when we are emiting a signal.
Emit the on_new_ssrc signal for the CSRC, not the SSRC.
Fixes #562319.
2009-08-11 02:30:39 +01:00
Wim Taymans
a80f7dc19a gst/rtpmanager/gstrtpbin.c: Reset the sync parameters when clearing the payload type map too.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_reset_sync),
(gst_rtp_bin_clear_pt_map):
Reset the sync parameters when clearing the payload type map too.
Fixes #562312.
2009-08-11 02:30:39 +01:00
Wim Taymans
a2d7487ee1 gst/rtpmanager/gstrtpbin.*: Remove a lot of per stream state that is not needed and pass new info in the method call.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (get_client),
(gst_rtp_bin_reset_sync), (gst_rtp_bin_associate),
(gst_rtp_bin_handle_sync), (create_stream),
(gst_rtp_bin_class_init), (new_ssrc_pad_found):
* gst/rtpmanager/gstrtpbin.h:
Remove a lot of per stream state that is not needed and pass new info in
the method call.
Add signal to reset sync parameters.
Avoid parsing the caps to get a clock_base, we get this from the sync
signal now.
2009-08-11 02:30:39 +01:00
Wim Taymans
b8408946b7 gst/rtpmanager/gstrtpsession.c: Fix event leak.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_event_send_rtcp_src):
Fix event leak.
2009-08-11 02:30:39 +01:00
Wim Taymans
ae346d9a6d gst/rtpmanager/rtpsession.c: Add property to configure the RTCP MTU.
Original commit message from CVS:
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (rtp_session_set_property),
(rtp_session_get_property):
Add property to configure the RTCP MTU.
2009-08-11 02:30:39 +01:00
Wim Taymans
55bb4d5c95 gst/rtpmanager/rtpsession.c: Add G_PARAM_STATIC_STRINGS.
Original commit message from CVS:
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(copy_source), (rtp_session_create_sources),
(rtp_session_get_property):
Add G_PARAM_STATIC_STRINGS.
Add property to return a GValueArray of all known RTPSources in the
session.
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_create_sdes), (rtp_source_set_property),
(rtp_source_get_property):
Remove properties to set the various SDES items, an application is never
supposed to change the RTPSource data.
Change the SDES getter properties to one SDES property that returns all
SDES items in a GstStructure.
2009-08-11 02:30:39 +01:00
Wim Taymans
c84ffd8460 gst/rtpmanager/gstrtpbin.c: Also unref the target pad for unknown pads.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_release_pad):
Also unref the target pad for unknown pads.
2009-08-11 02:30:39 +01:00
Olivier Crete
75580396d9 gst/rtpmanager/gstrtpbin.c: Release the right pads on rtpbin. Fixes #561752.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_release_pad):
Release the right pads on rtpbin. Fixes #561752.
2009-08-11 02:30:39 +01:00
Wim Taymans
2f5b130af3 gst/rtpmanager/gstrtpsession.c: Pass the running time to the session when processing RTP packets.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (get_current_times),
(rtcp_thread), (gst_rtp_session_chain_recv_rtp):
Pass the running time to the session when processing RTP packets.
Improve the time function to provide more info.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (update_arrival_stats),
(rtp_session_process_rtp), (rtp_session_process_sdes),
(rtp_session_process_rtcp), (session_start_rtcp),
(rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Mark the internal source with a flag.
Use running_time instead of the more useless timestamp.
Validate a source when a valid SDES has been received.
Pass the current system time when processing SR packets.
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_init), (rtp_source_create_stats),
(rtp_source_get_property), (rtp_source_send_rtp),
(rtp_source_process_rb), (rtp_source_get_new_rb),
(rtp_source_get_last_rb):
* gst/rtpmanager/rtpsource.h:
Add property to get source stats.
Mark params as STATIC_STRINGS.
Calculate the bitrate at the sender SSRC.
Avoid negative values in the round trip time calculations.
* gst/rtpmanager/rtpstats.h:
Update some docs and change some variable name to more closely reflect
what it contains.
2009-08-11 02:30:39 +01:00
Sebastian Dröge
e51423aab9 gst/rtpmanager/gstrtpjitterbuffer.c: Initialize return value to fix compiler warning about uninitialized variable.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain_rtcp):
Initialize return value to fix compiler warning about uninitialized
variable.
2009-08-11 02:30:39 +01:00
Wim Taymans
d0ada6127e gst/rtpmanager/gstrtpjitterbuffer.c: Mark signal arg as static scope.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init):
Mark signal arg as static scope.
2009-08-11 02:30:39 +01:00
Wim Taymans
592c3f222f gst/rtpmanager/gstrtpbin.c: Remove internal sync pad, use signals instead to get lip-sync notifications.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
(gst_rtp_bin_handle_sync), (create_stream), (free_stream),
(new_ssrc_pad_found):
Remove internal sync pad, use signals instead to get lip-sync
notifications.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_base_init),
(gst_rtp_jitter_buffer_class_init),
(gst_rtp_jitter_buffer_internal_links), (create_rtcp_sink),
(remove_rtcp_sink), (gst_rtp_jitter_buffer_request_new_pad),
(gst_rtp_jitter_buffer_release_pad),
(gst_rtp_jitter_buffer_sink_rtcp_event),
(gst_rtp_jitter_buffer_chain_rtcp),
(gst_rtp_jitter_buffer_get_property):
* gst/rtpmanager/gstrtpjitterbuffer.h:
Make it possible to send SR packets to the jitterbuffer.
Check if the SR timestamps are valid by comparing them to the RTP
timestamps.
Signal the SR packet and the timing information to listeners.
* gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc),
(gst_rtp_ssrc_demux_rtcp_chain), (gst_rtp_ssrc_demux_src_query):
Remove some unused code.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew), (rtp_jitter_buffer_get_sync):
* gst/rtpmanager/rtpjitterbuffer.h:
Keep track of the last seen RTP timestamp so that we can filter out
invalid SR packets.
2009-08-11 02:30:38 +01:00
Sebastian Dröge
c3645239f5 gst/rtpmanager/rtpsource.c: Fix GST_DEBUG call to only have as many arguments as required by the format string. Fixes...
Original commit message from CVS:
* gst/rtpmanager/rtpsource.c: (get_clock_rate):
Fix GST_DEBUG call to only have as many arguments as required
by the format string. Fixes a compiler warning.
2009-08-11 02:30:38 +01:00
Wim Taymans
5ab3e10594 gst/rtpmanager/gstrtpbin.c: Do not try to keep track of the clock-rate ourselves but simply get the value from the ji...
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
(gst_rtp_bin_sync_chain), (create_stream), (new_ssrc_pad_found):
Do not try to keep track of the clock-rate ourselves but simply get the
value from the jitterbuffer.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_get_sync):
* gst/rtpmanager/gstrtpjitterbuffer.h:
Add some debug info.
Pass the clock-rate to the jitterbuffer.
Also pass the clock-rate along with the rtp timestamp when getting the
sync parameters.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
Fix some debug.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew), (rtp_jitter_buffer_get_sync):
* gst/rtpmanager/rtpjitterbuffer.h:
Keep track of clock-rate changes and return the clock-rate together with
the rtp timestamps used for sync.
Don't try to construct timestamps when we have no base_time.
* gst/rtpmanager/rtpsource.c: (get_clock_rate):
Request a new clock-rate when the payload type changes.
Reset the jitter calculation when the clock-rate changes.
2009-08-11 02:30:38 +01:00
Wim Taymans
1656fad93e gst/rtpmanager/: Small cleanups and some more debug info.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_parse_caps),
(gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_chain):
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew):
Small cleanups and some more debug info.
2009-08-11 02:30:38 +01:00
Wim Taymans
6485d60a01 gst/rtpmanager/gstrtpjitterbuffer.c: Also configure the next expected output seqnum when we get a seqnum-base on the ...
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain):
Also configure the next expected output seqnum when we get a seqnum-base
on the caps.
2009-08-11 02:30:38 +01:00
Stefan Kost
b835296809 Don't install static libs for plugins. Fixes #550851 for -bad.
Original commit message from CVS:
* ext/alsaspdif/Makefile.am:
* ext/amrwb/Makefile.am:
* ext/apexsink/Makefile.am:
* ext/arts/Makefile.am:
* ext/artsd/Makefile.am:
* ext/audiofile/Makefile.am:
* ext/audioresample/Makefile.am:
* ext/bz2/Makefile.am:
* ext/cdaudio/Makefile.am:
* ext/celt/Makefile.am:
* ext/dc1394/Makefile.am:
* ext/dirac/Makefile.am:
* ext/directfb/Makefile.am:
* ext/divx/Makefile.am:
* ext/dts/Makefile.am:
* ext/faac/Makefile.am:
* ext/faad/Makefile.am:
* ext/gsm/Makefile.am:
* ext/hermes/Makefile.am:
* ext/ivorbis/Makefile.am:
* ext/jack/Makefile.am:
* ext/jp2k/Makefile.am:
* ext/ladspa/Makefile.am:
* ext/lcs/Makefile.am:
* ext/libfame/Makefile.am:
* ext/libmms/Makefile.am:
* ext/metadata/Makefile.am:
* ext/mpeg2enc/Makefile.am:
* ext/mplex/Makefile.am:
* ext/musepack/Makefile.am:
* ext/musicbrainz/Makefile.am:
* ext/mythtv/Makefile.am:
* ext/nas/Makefile.am:
* ext/neon/Makefile.am:
* ext/ofa/Makefile.am:
* ext/polyp/Makefile.am:
* ext/resindvd/Makefile.am:
* ext/sdl/Makefile.am:
* ext/shout/Makefile.am:
* ext/snapshot/Makefile.am:
* ext/sndfile/Makefile.am:
* ext/soundtouch/Makefile.am:
* ext/spc/Makefile.am:
* ext/swfdec/Makefile.am:
* ext/tarkin/Makefile.am:
* ext/theora/Makefile.am:
* ext/timidity/Makefile.am:
* ext/twolame/Makefile.am:
* ext/x264/Makefile.am:
* ext/xine/Makefile.am:
* ext/xvid/Makefile.am:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/dshow/Makefile.am:
* gst/aiffparse/Makefile.am:
* gst/app/Makefile.am:
* gst/audiobuffer/Makefile.am:
* gst/bayer/Makefile.am:
* gst/cdxaparse/Makefile.am:
* gst/chart/Makefile.am:
* gst/colorspace/Makefile.am:
* gst/dccp/Makefile.am:
* gst/deinterlace/Makefile.am:
* gst/deinterlace2/Makefile.am:
* gst/dvdspu/Makefile.am:
* gst/festival/Makefile.am:
* gst/filter/Makefile.am:
* gst/flacparse/Makefile.am:
* gst/flv/Makefile.am:
* gst/games/Makefile.am:
* gst/h264parse/Makefile.am:
* gst/librfb/Makefile.am:
* gst/mixmatrix/Makefile.am:
* gst/modplug/Makefile.am:
* gst/mpeg1sys/Makefile.am:
* gst/mpeg4videoparse/Makefile.am:
* gst/mpegdemux/Makefile.am:
* gst/mpegtsmux/Makefile.am:
* gst/mpegvideoparse/Makefile.am:
* gst/mve/Makefile.am:
* gst/nsf/Makefile.am:
* gst/nuvdemux/Makefile.am:
* gst/overlay/Makefile.am:
* gst/passthrough/Makefile.am:
* gst/pcapparse/Makefile.am:
* gst/playondemand/Makefile.am:
* gst/rawparse/Makefile.am:
* gst/real/Makefile.am:
* gst/rtjpeg/Makefile.am:
* gst/rtpmanager/Makefile.am:
* gst/scaletempo/Makefile.am:
* gst/sdp/Makefile.am:
* gst/selector/Makefile.am:
* gst/smooth/Makefile.am:
* gst/smoothwave/Makefile.am:
* gst/speed/Makefile.am:
* gst/speexresample/Makefile.am:
* gst/stereo/Makefile.am:
* gst/subenc/Makefile.am:
* gst/tta/Makefile.am:
* gst/vbidec/Makefile.am:
* gst/videodrop/Makefile.am:
* gst/videosignal/Makefile.am:
* gst/virtualdub/Makefile.am:
* gst/vmnc/Makefile.am:
* gst/y4m/Makefile.am:
* sys/acmenc/Makefile.am:
* sys/cdrom/Makefile.am:
* sys/dshowdecwrapper/Makefile.am:
* sys/dshowsrcwrapper/Makefile.am:
* sys/dvb/Makefile.am:
* sys/dxr3/Makefile.am:
* sys/fbdev/Makefile.am:
* sys/oss4/Makefile.am:
* sys/qcam/Makefile.am:
* sys/qtwrapper/Makefile.am:
* sys/vcd/Makefile.am:
* sys/wininet/Makefile.am:
* win32/common/config.h:
Don't install static libs for plugins. Fixes #550851 for -bad.
2009-08-11 02:30:38 +01:00
Wim Taymans
eaa23fd49a gst/rtpmanager/gstrtpjitterbuffer.c: Fix problem with using the output seqnum counter to check for input seqnum disco...
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_parse_caps),
(gst_rtp_jitter_buffer_flush_start),
(gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_loop):
Fix problem with using the output seqnum counter to check for input
seqnum discontinuities.
Improve gap detection and recovery, reset and flush the jitterbuffer on
seqnum restart. Fixes #556520.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_insert):
Fix wrong G_LIKELY.
2009-08-11 02:30:38 +01:00
Wim Taymans
3563bbaabd gst/rtpmanager/gstrtpsession.c: Install event handler on the rtcp_src pad, make LATENCY event return
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_event_send_rtcp_src), (create_send_rtcp_src):
Install event handler on the rtcp_src pad, make LATENCY event return
TRUE.
2009-08-11 02:30:38 +01:00
Håvard Graff
3bebd53b6f gst/rtpmanager/gstrtpbin-marshal.list: Add marshaller for new action signal.
Original commit message from CVS:
Patch by: Håvard Graff <havard dot graff at tandberg dot com>
* gst/rtpmanager/gstrtpbin-marshal.list:
Add marshaller for new action signal.
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_get_internal_session),
(gst_rtp_bin_class_init):
* gst/rtpmanager/gstrtpbin.h:
Add action signal to retrieve the internal RTPSession object.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_get_property), (gst_rtp_session_release_pad):
Add property to access the internal RTPSession object.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(check_collision):
* gst/rtpmanager/rtpsession.h:
Add action signal to retrieve an RTPSource object by SSRC.
See #555396.
2009-08-11 02:30:38 +01:00
Wim Taymans
bd8f4b6c58 gst/rtpmanager/gstrtpbin.c: Release pads of the session manager.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (find_session_by_pad),
(free_session), (gst_rtp_bin_dispose), (remove_recv_rtp),
(remove_recv_rtcp), (remove_send_rtp), (remove_rtcp),
(gst_rtp_bin_release_pad):
Release pads of the session manager.
Start implementing releasing pads of gstrtpbin.
* gst/rtpmanager/gstrtpsession.c: (remove_recv_rtp_sink),
(remove_recv_rtcp_sink), (remove_send_rtp_sink),
(remove_send_rtcp_src), (gst_rtp_session_release_pad):
Implement releasing pads in gstrtpsession.
2009-08-11 02:30:38 +01:00
Wim Taymans
4553863755 gst/rtpmanager/gstrtpjitterbuffer.c: Only update the seqnum-base when it was not already configured for the streams.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_parse_caps):
Only update the seqnum-base when it was not already configured for the
streams.
2009-08-11 02:30:37 +01:00
Wim Taymans
55b7860cc4 gst/rtpmanager/rtpsession.c: Ref the rtpsource object before we release the session lock when we emit the signals.
Original commit message from CVS:
* gst/rtpmanager/rtpsession.c: (on_new_ssrc), (on_ssrc_collision),
(on_ssrc_validated), (on_ssrc_active), (on_ssrc_sdes),
(on_bye_ssrc), (on_bye_timeout), (on_timeout), (on_sender_timeout):
Ref the rtpsource object before we release the session lock when we emit
the signals.
2009-08-11 02:30:37 +01:00
Wim Taymans
c2c69bfb86 gst/rtpmanager/: Fix some docs.
Original commit message from CVS:
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_insert),
(rtp_jitter_buffer_get_sync):
* gst/rtpmanager/rtpsession.c: (on_sender_timeout),
(session_cleanup):
* gst/rtpmanager/rtpsource.c:
Fix some docs.
2009-08-11 02:30:37 +01:00
Jan Schmidt
a2b86bbce5 Fix compiler warnings on OS/X
Original commit message from CVS:
* ext/jack/gstjackaudiosink.c: (jack_process_cb):
* gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew):
Fix compiler warnings on OS/X
2009-08-11 02:30:37 +01:00
Wim Taymans
5e98fa572f gst/rtpmanager/gstrtpbin.c: Do not try to adjust the offset of streams for which we have not yet seen an SR packet. A...
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session),
(gst_rtp_bin_associate), (gst_rtp_bin_sync_chain):
Do not try to adjust the offset of streams for which we have not yet
seen an SR packet. Avoids large ts-offsets in some cases.
2009-08-11 02:30:37 +01:00
Wim Taymans
85e26f6546 gst/rtpmanager/gstrtpbin.*: Add signal to notify listeners when a sender becomes a receiver.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (on_sender_timeout),
(create_session), (gst_rtp_bin_associate),
(gst_rtp_bin_sync_chain), (gst_rtp_bin_class_init),
(gst_rtp_bin_request_new_pad):
* gst/rtpmanager/gstrtpbin.h:
Add signal to notify listeners when a sender becomes a receiver.
Tweak lip-sync code, don't store our own copy of the ts-offset of the
jitterbuffer, don't adjust sync if the change is less than 4msec.
Get the RTP timestamp <-> GStreamer timestamp relation directly from
the jitterbuffer instead of our inaccurate version from the source.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_get_sync):
* gst/rtpmanager/gstrtpjitterbuffer.h:
Add G_LIKELY macros, use global defines for max packet reorder and
dropouts.
Reset the jitterbuffer clock skew detection when packets seqnums are
changed unexpectedly.
* gst/rtpmanager/gstrtpsession.c: (on_sender_timeout),
(gst_rtp_session_class_init), (gst_rtp_session_init):
* gst/rtpmanager/gstrtpsession.h:
Add sender timeout signal.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew), (rtp_jitter_buffer_insert),
(rtp_jitter_buffer_get_sync):
* gst/rtpmanager/rtpjitterbuffer.h:
Add some G_LIKELY macros.
Keep track of the extended RTP timestamp so that we can report the RTP
timestamp <-> GStreamer timestamp relation for lip-sync.
Remove server timestamp gap detection code, the server can sometimes
make a huge gap in timestamps (talk spurts,...) see #549774.
Detect timetamp weirdness instead by observing the sender/receiver
timestamp relation and resync if it changes more than 1 second.
Add method to report about the current rtp <-> gst timestamp relation
which is needed for lip-sync.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(on_sender_timeout), (check_collision), (rtp_session_process_sr),
(session_cleanup):
* gst/rtpmanager/rtpsession.h:
Add sender timeout signal.
Remove inaccurate rtp <-> gst timestamp relation code, the
jitterbuffer can now do an accurate reporting about this.
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(rtp_source_update_caps), (calculate_jitter),
(rtp_source_process_rtp):
* gst/rtpmanager/rtpsource.h:
Remove inaccurate rtp <-> gst timestamp relation code.
* gst/rtpmanager/rtpstats.h:
Define global max-reorder and max-dropout constants for use in various
subsystems.
2009-08-11 02:30:37 +01:00
Wim Taymans
5c89bb2ab3 gst/rtpmanager/gstrtpsession.c: Send EOS when the session object instructs us to.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_send_rtcp),
(gst_rtp_session_event_send_rtp_sink):
Send EOS when the session object instructs us to.
* gst/rtpmanager/rtpsession.c: (rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Make it possible for the session manager to instruct us to send EOS. We
currently will EOS when the session is a sender and when the sender part
goes EOS. This is not entirely correct behaviour because the session
could still participate as a receiver.
Fixes #549409.
2009-08-11 02:30:37 +01:00
Wim Taymans
62ecaee748 gst/rtpmanager/gstrtpbin.c: Reset rtp timestamp interpollation when we detect a gap when the clock_base changed.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
(gst_rtp_bin_sync_chain), (new_ssrc_pad_found):
Reset rtp timestamp interpollation when we detect a gap when the
clock_base changed.
Don't try to adjust the ts-offset when it's too big (> 3seconds)
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_set_ssrc):
* gst/rtpmanager/gstrtpsession.h:
Add method to set session SSRC.
* gst/rtpmanager/rtpsession.c: (check_collision),
(rtp_session_set_internal_ssrc), (rtp_session_get_internal_ssrc),
(rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Added debugging for the collision checks.
Add method to change the internal SSRC of the session.
* gst/rtpmanager/rtpsource.c: (rtp_source_process_rtp):
Reset the clock base when we detect large jumps in the seqnums.
2009-08-11 02:30:37 +01:00
Stefan Kost
cc74738d83 gst/rtpmanager/gstrtpbin.c: Print the pad-name in debug log.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c:
Print the pad-name in debug log.
* sys/dshowsrcwrapper/gstdshowaudiosrc.c:
* sys/dshowsrcwrapper/gstdshowvideosrc.c:
Use "-" instead of "_" in property names. Can we call them just
"device" like everywhere else?
2009-08-11 02:30:37 +01:00
Olivier Crete
d392defbd3 gst/rtpmanager/gstrtpjitterbuffer.c: Make the buffer metadata writable before inserting it in the jitterbuffer becaus...
Original commit message from CVS:
Based on patch by: Olivier Crete <tester at tester dot ca>
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
Make the buffer metadata writable before inserting it in the
jitterbuffer because the jitterbuffer will modify the timestamps.
* gst/rtpmanager/rtpjitterbuffer.c:
Update method comment about requiring writable metadata on buffers.
* gst/rtpmanager/rtpsession.c: (rtp_session_process_sr),
(rtp_session_process_rtcp):
Make the RTCP buffer metadata writable because we want to modify the
metadata.
Fixes #546312.
2009-08-11 02:30:37 +01:00
Håvard Graff
1bef5a8ab8 gst/rtpmanager/gstrtpjitterbuffer.c: Fix debug by logging the right seqnum.
Original commit message from CVS:
Patch by: Håvard Graff <havard dot graff at tandberg dot com>
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain):
Fix debug by logging the right seqnum.
2009-08-11 02:30:37 +01:00
Olivier Crete
2707a84d78 gst/rtpmanager/gstrtpbin.c: Release lock before emitting the request-pt-map signal.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtpmanager/gstrtpbin.c: (get_pt_map):
Release lock before emitting the request-pt-map signal.
Fixes #543480.
2009-08-11 02:30:37 +01:00
Peter Kjellerstedt
fd44690d4f gst/rtpmanager/: Corrected a typo (interpollate -> interpolate).
Original commit message from CVS:
* ChangeLog:
* gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_loop):
* gst/rtpmanager/rtpsource.c: (rtp_source_get_new_sr):
Corrected a typo (interpollate -> interpolate).
2009-08-11 02:30:36 +01:00
Peter Kjellerstedt
e2f49d9ccf gst/rtpmanager/: Changed some GST_DEBUG() to GST_LOG() to reduce the spam when a pipeline is running normally.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
(gst_rtp_session_send_rtp), (gst_rtp_session_send_rtcp),
(gst_rtp_session_sync_rtcp), (gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_chain_recv_rtcp), (gst_rtp_session_chain_send_rtp):
* gst/rtpmanager/rtpsession.c: (source_push_rtp),
(rtp_session_send_rtp):
* gst/rtpmanager/rtpsource.c: (push_packet), (calculate_jitter),
(rtp_source_process_rtp), (rtp_source_send_rtp):
Changed some GST_DEBUG() to GST_LOG() to reduce the spam when a
pipeline is running normally.
2009-08-11 02:30:36 +01:00
Peter Kjellerstedt
ca15984e14 gst/rtpmanager/: Do not mix the use of g_get_current_time() with gst_clock_get_time().
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_init),
(gst_rtp_session_finalize), (rtcp_thread),
(gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp):
* gst/rtpmanager/rtpsession.c: (check_collision),
(update_arrival_stats), (rtp_session_process_rtp),
(rtp_session_process_rtcp), (rtp_session_send_rtp),
(rtp_session_send_bye_locked), (rtp_session_send_bye),
(rtp_session_next_timeout), (session_report_blocks), (session_cleanup),
(is_rtcp_time), (rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Do not mix the use of g_get_current_time() with gst_clock_get_time().
2009-08-11 02:30:36 +01:00
Stefan Kost
a71ffc55d8 Final round of doc updates.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/speed/gstspeed.c:
* gst/speexresample/gstspeexresample.c:
* gst/videosignal/gstvideoanalyse.c:
* gst/videosignal/gstvideodetect.c:
* gst/videosignal/gstvideomark.c:
* sys/dvb/gstdvbsrc.c:
* sys/oss4/oss4-mixer.c:
* sys/oss4/oss4-sink.c:
* sys/oss4/oss4-source.c:
* sys/wininet/gstwininetsrc.c:
Final round of doc updates.
2009-08-11 02:30:36 +01:00
Stefan Kost
138c2b7cf9 gst/: More doc updates. More xrefs.
Original commit message from CVS:
* gst/deinterlace/gstdeinterlace.c:
* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/gstrtpssrcdemux.c:
* gst/sdp/gstsdpdemux.c:
More doc updates. More xrefs.
2009-08-11 02:30:36 +01:00
Stefan Kost
2d1ccbf52e Do not use short_description in section docs for elements. We extract them from element details and there will be war...
Original commit message from CVS:
* ext/dc1394/gstdc1394.c:
* ext/ivorbis/vorbisdec.c:
* ext/jack/gstjackaudiosink.c:
* ext/metadata/gstmetadatademux.c:
* ext/mythtv/gstmythtvsrc.c:
* ext/theora/theoradec.c:
* gst-libs/gst/app/gstappsink.c:
* gst/bayer/gstbayer2rgb.c:
* gst/deinterlace/gstdeinterlace.c:
* gst/rawparse/gstaudioparse.c:
* gst/rawparse/gstvideoparse.c:
* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/gstrtpssrcdemux.c:
* gst/selector/gstinputselector.c:
* gst/selector/gstoutputselector.c:
* gst/videosignal/gstvideoanalyse.c:
* gst/videosignal/gstvideodetect.c:
* gst/videosignal/gstvideomark.c:
* sys/oss4/oss4-mixer.c:
* sys/oss4/oss4-sink.c:
* sys/oss4/oss4-source.c:
Do not use short_description in section docs for elements. We extract
them from element details and there will be warnings if they differ.
Also fixing up the ChangeLog order.
2009-08-11 02:30:36 +01:00
Wim Taymans
8dc879f15e gst/rtpmanager/gstrtpbin.c: Fix deadlock when shutting down, use a new lock instead to properly shutdown.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_init),
(gst_rtp_bin_finalize), (gst_rtp_bin_change_state):
Fix deadlock when shutting down, use a new lock instead to properly
shutdown.
2009-08-11 02:30:36 +01:00
Wim Taymans
fda8195d76 gst/rtpmanager/gstrtpbin.c: Break out of callbacks when we are shutting down.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c:
(gst_rtp_bin_propagate_property_to_jitterbuffer),
(gst_rtp_bin_change_state), (new_payload_found),
(new_ssrc_pad_found):
Break out of callbacks when we are shutting down.
Make sure no state changes can happen when we reconfigure.
2009-08-11 02:30:36 +01:00
Wim Taymans
bd1e0ebfc0 gst/rtpmanager/gstrtpjitterbuffer.c: When checking the seqnum, reset the jitterbuffer if the gap is too big, we need ...
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
When checking the seqnum, reset the jitterbuffer if the gap is too big,
we need to do this so that we can better handle a restarted source.
Fix some comments.
* gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew),
(rtp_jitter_buffer_insert):
Tweak the skew resync diff.
Use our working seqnum compare function in -base.
Rework the jitterbuffer insert code to make it clearer and more
performant by only retrieving the seqnum of the input buffer once and by
adding some G_LIKELY compiler hints.
Improve debugging for duplicate packets.
* gst/rtpmanager/rtpsource.c: (rtp_source_process_rtp):
Fix a comment, we don't do skew correction here..
2009-08-11 02:30:36 +01:00
Håvard Graff
b889dfad30 gst/rtpmanager/gstrtpbin.c: Propagate the do-lost and latency properties to the jitterbuffers when they are changed o...
Original commit message from CVS:
Patch by: Håvard Graff <havard dot graff at tandberg dot com>
* gst/rtpmanager/gstrtpbin.c:
(gst_rtp_bin_propagate_property_to_jitterbuffer),
(gst_rtp_bin_set_property):
Propagate the do-lost and latency properties to the jitterbuffers when
they are changed on rtpbin.
2009-08-11 02:30:36 +01:00
Wim Taymans
6716231857 Don't use _gst_pad().
Original commit message from CVS:
* examples/switch/switcher.c: (switch_timer):
* gst/replaygain/gstrgvolume.c: (gst_rg_volume_init):
* gst/rtpmanager/gstrtpclient.c: (create_stream):
* gst/sdp/gstsdpdemux.c: (gst_sdp_demux_stream_configure_udp),
(gst_sdp_demux_stream_configure_udp_sink):
* tests/check/elements/deinterleave.c: (GST_START_TEST),
(pad_added_setup_data_check_float32_8ch_cb):
* tests/check/elements/rganalysis.c: (send_eos_event),
(send_tag_event):
Don't use _gst_pad().
2009-08-11 02:30:35 +01:00
Jan Schmidt
4e5347c8fe docs/Makefile.am: Don't attempt to build plugin docs when they're disabled.
Original commit message from CVS:
* docs/Makefile.am:
Don't attempt to build plugin docs when they're disabled.
* gst/bayer/Makefile.am:
Add libgstvideo to the link.
* gst/rtpmanager/Makefile.am:
Fix link order, and move LIBS things to _LIBS
2009-08-11 02:30:35 +01:00
Wim Taymans
2506d13ecc gst/rtpmanager/gstrtpjitterbuffer.c: Simply drop bad RTP packets with a warning instead of just posting an error and ...
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain):
Simply drop bad RTP packets with a warning instead of just posting an
error and stopping. This is a perfectly recoverable event and we don't
force people to use an rtpbin to filter out bad packets first.
2009-08-11 02:30:35 +01:00
Wim Taymans
cd00eb71b4 gst/rtpmanager/gstrtpbin.c: Actually add the do-lost property to the object.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init):
Actually add the do-lost property to the object.
2009-08-11 02:30:35 +01:00
Wim Taymans
71c2510665 gst/rtpmanager/gstrtpjitterbuffer.c: Avoid waiting for a negative (huge) duration when the last packet has a lower ti...
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_loop):
Avoid waiting for a negative (huge) duration when the last packet has a
lower timestamp than the current packet.
2009-08-11 02:30:35 +01:00
Peter Kjellerstedt
fd8061784a gst/rtpmanager/gstrtpsession.c: Make sure to unref the rtpsession returned by gst_pad_get_parent() to prevent a memor...
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_query_send_rtcp_src):
Make sure to unref the rtpsession returned by gst_pad_get_parent() to
prevent a memory leak.
2009-08-11 02:30:35 +01:00
Jan Schmidt
95ab282083 gst/rtpmanager/gstrtpjitterbuffer.c: Initialise with GST_CLOCK_TIME_NONE to avoid compiler warning.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_loop):
Initialise with GST_CLOCK_TIME_NONE to avoid compiler warning.
2009-08-11 02:30:35 +01:00
Peter Kjellerstedt
b1ef03968a gst/rtpmanager/rtpsource.c: Make sure to unref the caps used by RTPSource to prevent a memory leak.
Original commit message from CVS:
* gst/rtpmanager/rtpsource.c: (rtp_source_finalize):
Make sure to unref the caps used by RTPSource to prevent a memory leak.
2009-08-11 02:30:35 +01:00
Olivier Crete
bddddbd409 gst/rtpmanager/rtpsession.c: Unlock the session lock when calling one of our callbacks.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtpmanager/rtpsession.c: (source_clock_rate),
(rtp_session_process_bye), (rtp_session_send_bye_locked):
Unlock the session lock when calling one of our callbacks.
Fixes #532011.
2009-08-11 02:30:35 +01:00
Sjoerd Simons
c466ae6bdc gst/rtpmanager/gstrtpsession.c: Send RTP BYE command on EOS. Fixes bug #531955.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_event_send_rtp_sink):
Send RTP BYE command on EOS. Fixes bug #531955.
2009-08-11 02:30:35 +01:00
Wim Taymans
d6c8809739 gst/rtpmanager/gstrtpbin.*: Expose new jitterbuffer property in rtpbin too.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_stream), (gst_rtp_bin_init),
(gst_rtp_bin_set_property), (gst_rtp_bin_get_property):
* gst/rtpmanager/gstrtpbin.h:
Expose new jitterbuffer property in rtpbin too.
2009-08-11 02:30:35 +01:00
Wim Taymans
250c38a5ce gst/rtpmanager/gstrtpjitterbuffer.c: Disable sending out rtp packet lost events by default and make a property to ena...
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_init),
(gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_set_property),
(gst_rtp_jitter_buffer_get_property):
Disable sending out rtp packet lost events by default and make a
property to enabe it. We will likely enable it by default when the base
depayloaders have a default handler for them so that we don't send these
events all through the pipeline for now.
2009-08-11 02:30:35 +01:00
Wim Taymans
e2ab966d14 gst/rtpmanager/gstrtpjitterbuffer.c: Remove private version of a function that is in -base now.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_flush_stop),
(gst_rtp_jitter_buffer_src_event), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_loop):
Remove private version of a function that is in -base now.
Add src event handler.
Rework the jitterbuffer pushing loop so that it can quickly react to
lost packets and instruct the depayloader of them. This can then be used
to implement error concealment data.
2009-08-11 02:30:34 +01:00
Wim Taymans
a05b42ef04 gst/rtpmanager/gstrtpsession.c: Set up some internal links functions for the RTCP and sync pads because the defaults ...
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_query_send_rtcp_src), (create_recv_rtcp_sink),
(create_send_rtcp_src):
Set up some internal links functions for the RTCP and sync pads because
the defaults are really not correct.
Implement a query handler for the RTCP src pad, mostly to correctly
report about the latency.
2009-08-11 02:30:34 +01:00
Wim Taymans
e779adca69 gst/rtpmanager/: Also keep track of the first buffer timestamp together with the first
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
(gst_rtp_bin_sync_chain):
* gst/rtpmanager/rtpsession.c: (update_arrival_stats),
(rtp_session_process_sr), (rtp_session_on_timeout):
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(calculate_jitter):
* gst/rtpmanager/rtpsource.h:
* gst/rtpmanager/rtpstats.h:
Also keep track of the first buffer timestamp together with the first
RTP timestamp as they both are needed to construct the timing of
outgoing packets in the jitterbuffer and are therefore also needed to
manage lip-sync. This fixes lip-sync if the first RTP packets arrive
with a wildly different gap.
2009-08-11 02:30:34 +01:00
Olivier Crete
3c5cf0cd38 gst/rtpmanager/gstrtpbin.c: Ref caps when inserting into the cache.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
(new_ssrc_pad_found):
Ref caps when inserting into the cache.
Don't leak pads.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_get_clock_rate),
(gst_rtp_jitter_buffer_query):
Avoid a caps leak.
Don't leak refcount in query.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_get_caps),
(gst_rtp_pt_demux_chain):
Avoid caps leaks.
* gst/rtpmanager/gstrtpsession.c: (source_get_sdes_structure),
(gst_rtp_session_init), (return_true),
(gst_rtp_session_clear_pt_map), (gst_rtp_session_cache_caps),
(gst_rtp_session_clock_rate):
Ref caps when inserting into the cache.
Fix some more caps leaks. Fixes #528245.
2009-08-11 02:30:34 +01:00
Wim Taymans
4cc70a0c22 gst/rtpmanager/: Unset GValues after g_signal_emitv so that we avoid a refcount leak.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (get_pt_map), (free_client),
(gst_rtp_bin_associate), (gst_rtp_bin_get_free_pad_name):
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_get_clock_rate):
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_get_caps):
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_clock_rate):
Unset GValues after g_signal_emitv so that we avoid a refcount leak.
Don't leak a padname.
Don't leak client streams list.
Lock rtpbin when associating streams. Fixes #528245.
2009-08-11 02:30:34 +01:00
Peter Kjellerstedt
959c341cbd gst/rtpmanager/: Avoid leaking pads in the RTP manager.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (free_session):
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_finalize):
Avoid leaking pads in the RTP manager.
2009-08-11 02:30:34 +01:00
Olivier Crete
3f58847080 gst/rtpmanager/rtpsession.*: Implement collision and loop detection in rtpmanager.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtpmanager/rtpsession.c: (find_add_conflicting_addresses),
(check_collision), (obtain_source), (rtp_session_create_new_ssrc),
(rtp_session_create_source), (rtp_session_process_rtp),
(rtp_session_process_sr), (rtp_session_process_rr),
(rtp_session_process_sdes), (rtp_session_process_bye),
(rtp_session_send_bye_locked), (rtp_session_send_bye),
(rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Implement collision and loop detection in rtpmanager.
Fixes #520626.
* gst/rtpmanager/rtpsource.c: (rtp_source_reset),
(rtp_source_init):
* gst/rtpmanager/rtpsource.h:
Add method to reset stats.
2009-08-11 02:30:34 +01:00
Ole André Vadla Ravnås
6ba2fcd4ff gst/rtpmanager/gstrtpsession.c: Avoid a deadlock when joining the RTCP thread in PAUSED because it might be blocked d...
Original commit message from CVS:
Based on patch by: Ole André Vadla Ravnås  <ole.andre.ravnas@tandberg.com>
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_init),
(rtcp_thread), (start_rtcp_thread), (stop_rtcp_thread),
(join_rtcp_thread), (gst_rtp_session_change_state):
Avoid a deadlock when joining the RTCP thread in PAUSED because it might
be blocked downstream. Also avoid spawning multiple rtcp threads.
Fixes #520894.
2009-08-11 02:30:34 +01:00
Stefan Kost
52cdd3c59a gst/rtpmanager/rtpjitterbuffer.c: Don't try to reset the clock skew when we have no timestamps.
Original commit message from CVS:
Patch by: Stefan Kost <ensonic@users.sf.net>
* gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew):
Don't try to reset the clock skew when we have no timestamps.
Fixes #519005.
2009-08-11 02:30:34 +01:00
Olivier Crete
db8bdc8b92 gst/rtpmanager/gstrtpbin.c: Fix small memory leak, leaking caps. Fixes #bug 517571.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtpmanager/gstrtpbin.c: (new_ssrc_pad_found):
Fix small memory leak, leaking caps. Fixes #bug 517571.
2009-08-11 02:30:34 +01:00
Olivier Crete
a301c9a22b gst/rtpmanager/gstrtpbin.c: Ignore streams that did not receive an SR packet when doing synchronisation. Fixes #516160.
Original commit message from CVS:
Patch by: Olivier Crete <tester@tester.ca>
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate):
Ignore streams that did not receive an SR packet when doing
synchronisation. Fixes #516160.
2009-08-11 02:30:34 +01:00
Thijs Vermeir
b638626053 gst/rtpmanager/gstrtpjitterbuffer.c: Try to get the new clock-rate from the buffer caps when we receive a new payload...
Original commit message from CVS:
Patch by: Thijs Vermeir  <thijsvermeir at gmail dot com>
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain):
Try to get the new clock-rate from the buffer caps when we receive a new
payload type instead of always firing the signal. Fixes #512774.
2009-08-11 02:30:33 +01:00
Olivier Crete
7b2446b676 gst/rtpmanager/gstrtpbin.c: Also handle lip-sync when the clock-rate is not provided with caps but with a signal.
Original commit message from CVS:
Patch by: Olivier Crete <tester@tester.ca>
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
(create_stream), (payload_type_change), (new_ssrc_pad_found):
Also handle lip-sync when the clock-rate is not provided with caps but
with a signal.
2009-08-11 02:30:33 +01:00
Olivier Crete
41ada27f2e gst/rtpmanager/: Remove the fixed clock-rate from the jitterbuffer and extend it so that a clock-rate can be provided...
Original commit message from CVS:
Patch by: Olivier Crete <tester@tester.ca>
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain):
* gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew),
(rtp_jitter_buffer_insert):
* gst/rtpmanager/rtpjitterbuffer.h:
Remove the fixed clock-rate from the jitterbuffer and extend it so that
a clock-rate can be provided with each buffer instead. Fixes #511686.
2009-08-11 02:30:33 +01:00
Olivier Crete
eb0993af12 gst/rtpmanager/gstrtpjitterbuffer.c: Remove old unused variable.
Original commit message from CVS:
Patch by: Olivier Crete <tester@tester.ca>
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_change_state),
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
Remove old unused variable.
Track pt on input buffers and get the clock-rate when it changes.
Ignore packets with unknown clock-rate. See #511686.
2009-08-11 02:30:33 +01:00
Olivier Crete
0369f87020 gst/rtpmanager/rtpsource.c: Fix unref of buffer using the wrong function. Fixes #511920
Original commit message from CVS:
Patch by: Olivier Crete <tester@tester.ca>
* gst/rtpmanager/rtpsource.c: Fix unref of buffer using the
wrong function.  Fixes #511920
2009-08-11 02:30:33 +01:00
Wim Taymans
6e6c59a198 gst/rtpmanager/gstrtpsession.c: If we find the caps in the cache, use it to parse the clock-rate instead of returning...
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_clock_rate):
If we find the caps in the cache, use it to parse the clock-rate instead
of returning an error. Fixes a TODO as found by Youness Alaoui.
2009-08-11 02:30:33 +01:00
Youness Alaoui
03d9faf5fa gst/rtpmanager/: Make it possible to use different user_data for each of the callbacks.
Original commit message from CVS:
Patch by: Youness Alaoui <youness dot alaoui at collabora dot co dot uk>
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_clock_rate):
* gst/rtpmanager/rtpsession.c: (rtp_session_set_callbacks),
(rtp_session_set_process_rtp_callback),
(rtp_session_set_send_rtp_callback),
(rtp_session_set_send_rtcp_callback),
(rtp_session_set_sync_rtcp_callback),
(rtp_session_set_clock_rate_callback),
(rtp_session_set_reconsider_callback), (source_push_rtp),
(source_clock_rate), (rtp_session_process_bye),
(rtp_session_process_rtcp), (rtp_session_send_bye),
(rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Make it possible to use different user_data for each of the callbacks.
Fixes #508587.
2009-08-11 02:30:33 +01:00
Thijs Vermeir
c6d892420a gst/rtpmanager/gstrtpbin.c: Fix documentation for latest patch
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c:
Fix documentation for latest patch
2009-08-11 02:30:33 +01:00
Thijs Vermeir
a4db9d0943 gst/rtpmanager/gstrtpbin.c: Allow request_new_pad with name NULL (bug #508515)
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c:
Allow request_new_pad with name NULL (bug #508515)
2009-08-11 02:30:33 +01:00
Wim Taymans
c7818b0c0f gst/rtpmanager/gstrtpsession.c: Don't set fixed caps, we can basically do everything the upsteam peer pad can renegot...
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (create_send_rtp_sink):
Don't set fixed caps, we can basically do everything the upsteam peer
pad can renegotiate to. Fixes #507940.
2009-08-11 02:30:33 +01:00
Wim Taymans
c5e9700eda gst/rtpmanager/gstrtpjitterbuffer.c: Don't unref the popped buffer when we don't have ownership.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_loop):
Don't unref the popped buffer when we don't have ownership.
Fixes #507020.
2009-08-11 02:30:33 +01:00
Wim Taymans
cba910a430 gst/rtpmanager/gstrtpssrcdemux.c: Don't clean up pads when going to PAUSED.
Original commit message from CVS:
* gst/rtpmanager/gstrtpssrcdemux.c:
(gst_rtp_ssrc_demux_change_state):
Don't clean up pads when going to PAUSED.
2009-08-11 02:30:32 +01:00
Wim Taymans
a965ebff09 gst/rtpmanager/: Clean up the dynamic pads when going to READY.
Original commit message from CVS:
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_finalize),
(gst_rtp_pt_demux_setup), (gst_rtp_pt_demux_release),
(gst_rtp_pt_demux_change_state):
* gst/rtpmanager/gstrtpssrcdemux.c: (gst_rtp_ssrc_demux_reset),
(gst_rtp_ssrc_demux_dispose), (gst_rtp_ssrc_demux_src_query),
(gst_rtp_ssrc_demux_change_state):
Clean up the dynamic pads when going to READY.
2009-08-11 02:30:32 +01:00
Wim Taymans
df55cf2f08 gst/rtpmanager/: Fix some leaks.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_finalize),
(gst_rtp_bin_set_sdes_string), (gst_rtp_bin_get_sdes_string),
(gst_rtp_bin_handle_message):
* gst/rtpmanager/rtpsession.c: (rtp_session_finalize),
(rtp_session_send_bye):
* gst/rtpmanager/rtpsource.c: (rtp_source_finalize):
Fix some leaks.
2009-08-11 02:30:32 +01:00
Wim Taymans
771ed2339d gst/rtpmanager/: Post a message when the SDES infor changes for a source.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init),
(gst_rtp_bin_handle_message):
* gst/rtpmanager/gstrtpsession.c: (source_get_sdes_structure),
(on_ssrc_sdes):
Post a message when the SDES infor changes for a source.
* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpsource.c:
Update some comments.
2009-08-11 02:30:32 +01:00
Wim Taymans
49e501a647 gst/rtpmanager/: Add signal to notify of an SDES change.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (on_ssrc_sdes), (create_session),
(gst_rtp_bin_class_init):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpclient.h:
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpmanager.c:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpptdemux.h:
* gst/rtpmanager/gstrtpsession.c: (on_ssrc_sdes),
(gst_rtp_session_class_init), (gst_rtp_session_init):
* gst/rtpmanager/gstrtpsession.h:
* gst/rtpmanager/gstrtpssrcdemux.c:
* gst/rtpmanager/gstrtpssrcdemux.h:
* gst/rtpmanager/rtpjitterbuffer.c:
* gst/rtpmanager/rtpjitterbuffer.h:
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(on_ssrc_sdes), (rtp_session_process_sdes):
* gst/rtpmanager/rtpsession.h:
* gst/rtpmanager/rtpsource.c:
* gst/rtpmanager/rtpsource.h:
* gst/rtpmanager/rtpstats.c:
* gst/rtpmanager/rtpstats.h:
Add signal to notify of an SDES change.
Fix object type in the signal callbacks.
2009-08-11 02:30:32 +01:00
Wim Taymans
95d1f62397 gst/rtpmanager/gstrtpbin.*: Expose SDES items as properties and configure the session managers with them.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session),
(gst_rtp_bin_class_init), (gst_rtp_bin_init), (sdes_type_to_name),
(gst_rtp_bin_set_sdes_string), (gst_rtp_bin_get_sdes_string),
(gst_rtp_bin_set_property), (gst_rtp_bin_get_property):
* gst/rtpmanager/gstrtpbin.h:
Expose SDES items as properties and configure the session managers with
them.
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_set_property):
Fix SSRC property.
2009-08-11 02:30:32 +01:00
Wim Taymans
1971ae0d82 gst/rtpmanager/: Update comment.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session):
* gst/rtpmanager/rtpjitterbuffer.c:
Update comment.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_set_property), (gst_rtp_session_get_property):
Define some GObject properties to set SDES and other configuration.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (rtp_session_finalize),
(rtp_session_set_property), (rtp_session_get_property),
(on_ssrc_sdes), (rtp_session_set_bandwidth),
(rtp_session_get_bandwidth), (rtp_session_set_rtcp_fraction),
(rtp_session_get_rtcp_fraction), (rtp_session_set_sdes_string),
(rtp_session_get_sdes_string), (obtain_source),
(rtp_session_get_internal_source), (rtp_session_process_sdes),
(rtp_session_send_rtp), (rtp_session_next_timeout), (session_sdes),
(is_rtcp_time):
* gst/rtpmanager/rtpsession.h:
Add signal when new SDES infor has been found for a source.
Create properties for SDES and other info.
Simplify the SDES API.
Add method for getting the internal source object of the session.
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_finalize), (rtp_source_set_property),
(rtp_source_get_property), (rtp_source_set_callbacks),
(rtp_source_get_ssrc), (rtp_source_set_as_csrc),
(rtp_source_is_as_csrc), (rtp_source_is_active),
(rtp_source_is_validated), (rtp_source_is_sender),
(rtp_source_received_bye), (rtp_source_get_bye_reason),
(rtp_source_set_sdes), (rtp_source_set_sdes_string),
(rtp_source_get_sdes), (rtp_source_get_sdes_string),
(rtp_source_get_new_sr), (rtp_source_get_new_rb):
* gst/rtpmanager/rtpsource.h:
Add GObject properties for various things.
Don't leak the bye reason.
2009-08-11 02:30:32 +01:00
Wim Taymans
1a8f489093 gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer can buffer an unlimited amount of time and thus has no max_latency ...
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_query):
jitterbuffer can buffer an unlimited amount of time and thus has no
max_latency requirements.
2009-08-11 02:30:32 +01:00
Ole André Vadla Ravnås
c5fdb6bff3 gst/rtpmanager/gstrtpsession.c: Fix bad function signatures (#492798).
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås  <ole.andre.ravnas@tandberg.com>
* gst/rtpmanager/gstrtpsession.c:
Fix bad function signatures (#492798).
2009-08-11 02:30:32 +01:00
Laurent Glayal
8da59edc68 gst/rtpmanager/gstrtpbin.c: Fix memleak. Fixes #484990.
Original commit message from CVS:
Patch by: Laurent Glayal <spglegle at yahoo dot fr>
* gst/rtpmanager/gstrtpbin.c: (create_stream),
(gst_rtp_bin_class_init):
Fix memleak. Fixes #484990.
2009-08-11 02:30:31 +01:00
Jan Schmidt
c924d4a466 gst/: Fix compiler warnings shown by Forte.
Original commit message from CVS:
* gst/librfb/rfbbuffer.c: (rfb_buffer_new_and_alloc):
* gst/librfb/rfbbuffer.h:
* gst/librfb/rfbdecoder.c: (rfb_socket_get_buffer):
* gst/mpegvideoparse/mpegvideoparse.c: (gst_mpegvideoparse_chain):
* gst/nsf/nes6502.c: (nes6502_execute):
* gst/real/gstrealaudiodec.c: (gst_real_audio_dec_setcaps):
* gst/real/gstrealvideodec.c: (open_library):
* gst/real/gstrealvideodec.h:
* gst/rtpmanager/gstrtpsession.c: (create_recv_rtp_sink),
(create_recv_rtcp_sink), (create_send_rtp_sink):
Fix compiler warnings shown by Forte.
2009-08-11 02:30:31 +01:00
Wim Taymans
4556ccb666 gst/rtpmanager/gstrtpbin.c: Fix caps refcounting for payload maps.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (get_pt_map),
(gst_rtp_bin_clear_pt_map), (gst_rtp_bin_class_init):
Fix caps refcounting for payload maps.
When clearing payload maps, also clear sessions and streams payload
maps.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_get_caps),
(gst_rtp_pt_demux_clear_pt_map), (gst_rtp_pt_demux_chain),
(find_pad_for_pt):
Implement clearing the payload map.
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_event_send_rtp_sink):
Forward flush events instead of leaking them.
* gst/rtpmanager/gstrtpssrcdemux.c:
(gst_rtp_ssrc_demux_rtcp_sink_event):
Correctly refcount events before pushing them.
2009-08-11 02:30:31 +01:00
Wim Taymans
76a89b5e50 gst/rtpmanager/rtpsession.c: When reconsidering RTCP timeouts, set the next timeout against the last report time inst...
Original commit message from CVS:
* gst/rtpmanager/rtpsession.c: (rtp_session_next_timeout),
When reconsidering RTCP timeouts, set the next timeout against the last
report time instead of the current clock time so that we don't end up
reconsidering forever.
2009-08-11 02:30:31 +01:00
Wim Taymans
387f41e157 gst/rtpmanager/gstrtpjitterbuffer.c: Only peek at the tail element instead of popping it off, which allows us to grea...
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
Only peek at the tail element instead of popping it off, which allows
us to greatly simplify things when the tail element changes.
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_event_recv_rtp_sink):
* gst/rtpmanager/gstrtpssrcdemux.c:
(gst_rtp_ssrc_demux_sink_event):
Forward FLUSH events instead of leaking them.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew), (rtp_jitter_buffer_insert):
* gst/rtpmanager/rtpjitterbuffer.h:
Remove the tail-changed callback in favour of a simple boolean when we
insert a buffer in the queue.
Add method to peek the tail of the buffer.
2009-08-11 02:30:31 +01:00
Wim Taymans
b09507ab0c gst/rtpmanager/gstrtpjitterbuffer.c: Remove some old unused variables.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_flush_start),
(gst_rtp_jitter_buffer_flush_stop),
(gst_rtp_jitter_buffer_change_state), (apply_offset),
(gst_rtp_jitter_buffer_loop):
Remove some old unused variables.
Don't add the latency to the skew corrected timestamp, latency is only
used to sync against the clock.
Improve debugging.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init),
(rtp_jitter_buffer_reset_skew), (calculate_skew):
* gst/rtpmanager/rtpjitterbuffer.h:
Handle case where server timestamp goes backwards or wildly jumps by
temporarily pausing the skew correction.
Improve debugging.
2009-08-11 02:30:30 +01:00
Wim Taymans
9c867a2160 gst/rtpmanager/gstrtpbin.c: Fix crasher in dispose.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (free_client):
Fix crasher in dispose.
* gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew):
Handle cases where input buffers have no timestamps so that no clock
skew can be calculated, in this case interpollate timestamps based on
rtp timestamp and assume a 0 clock skew.
2009-08-11 02:30:30 +01:00
Wim Taymans
2b1f49a26e gst/rtpmanager/gstrtpjitterbuffer.c: Remove jitter correction code, it's now in the lower level object.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c: (apply_latency),
(gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query):
Remove jitter correction code, it's now in the lower level object.
Use new -core method for doing a peer query.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init),
(calculate_skew), (rtp_jitter_buffer_insert):
* gst/rtpmanager/rtpjitterbuffer.h:
Move jitter correction to the lowlevel jitterbuffer.
Increase the max window size.
When filling the window, already start estimating the skew using a
parabolic weighting factor so that we have a much better startup
behaviour that gets more accurate with the more samples we have.
Increase the default weighting factor for the steady state to get
smoother timestamps.
2009-08-11 02:30:30 +01:00
Wim Taymans
fa00695a39 gst/rtpmanager/gstrtpbin.c: Fix cleanup crasher.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_dispose),
(gst_rtp_bin_finalize):
Fix cleanup crasher.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init),
(calculate_skew):
* gst/rtpmanager/rtpjitterbuffer.h:
Dynamically adjust the skew calculation window so that we calculate it
over a period of around 2 seconds.
2009-08-11 02:30:30 +01:00
Wim Taymans
949f1685ce gst/rtpmanager/: Add notification of active SSRCs to various RTP elements. Fixes #478566.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (on_ssrc_active), (create_session),
(gst_rtp_bin_class_init):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpsession.c: (on_ssrc_active),
(gst_rtp_session_class_init), (gst_rtp_session_init),
(gst_rtp_session_event_send_rtp_sink):
* gst/rtpmanager/gstrtpsession.h:
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(on_ssrc_active), (rtp_session_process_rb):
* gst/rtpmanager/rtpsession.h:
Add notification of active SSRCs to various RTP elements. Fixes #478566.
2009-08-11 02:30:30 +01:00
Wim Taymans
56d5832287 gst/rtpmanager/gstrtpbin.c: Link to the right pads regardless of which one was created first in the ssrc demuxer.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (new_ssrc_pad_found):
Link to the right pads regardless of which one was created first in the
ssrc demuxer.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
(gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_send_rtp):
* gst/rtpmanager/rtpsource.c: (calculate_jitter):
Improve debugging.
* gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc),
(gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_finalize),
(gst_rtp_ssrc_demux_sink_event),
(gst_rtp_ssrc_demux_rtcp_sink_event), (gst_rtp_ssrc_demux_chain),
(gst_rtp_ssrc_demux_rtcp_chain),
(gst_rtp_ssrc_demux_internal_links):
* gst/rtpmanager/gstrtpssrcdemux.h:
Fix race in creating the RTP and RTCP pads when a new SSRC is detected.
2009-08-11 02:30:30 +01:00
Wim Taymans
b2aa36cb0d gst/rtpmanager/gstrtpbin.c: Use lock to protect variable.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_set_property),
(gst_rtp_bin_get_property):
Use lock to protect variable.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain),
(convert_rtptime_to_gsttime), (gst_rtp_jitter_buffer_loop):
Reconstruct GST timestamp from RTP timestamps based on measured clock
skew and sync offset.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init),
(rtp_jitter_buffer_set_tail_changed),
(rtp_jitter_buffer_set_clock_rate),
(rtp_jitter_buffer_get_clock_rate), (calculate_skew),
(rtp_jitter_buffer_insert), (rtp_jitter_buffer_peek):
* gst/rtpmanager/rtpjitterbuffer.h:
Measure clock skew.
Add callback to be notfied when a new packet was inserted at the tail.
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(calculate_jitter), (rtp_source_send_rtp):
* gst/rtpmanager/rtpsource.h:
Remove clock skew detection, it's move to the jitterbuffer now.
2009-08-11 02:30:30 +01:00
Wim Taymans
0441ef80b0 gst/rtpmanager/gstrtpbin.c: Also set NTP base time on new sessions.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session):
Also set NTP base time on new sessions.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query),
(gst_rtp_jitter_buffer_set_property),
(gst_rtp_jitter_buffer_get_property):
Use the right lock to protect our variables.
Fix some comment.
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_getcaps_send_rtp),
(gst_rtp_session_chain_send_rtp), (create_send_rtp_sink):
Implement getcaps on the sender sinkpad so that payloaders can negotiate
the right SSRC.
2009-08-11 02:30:30 +01:00
Wim Taymans
a93348cc6d gst/rtpmanager/: Various leak fixes.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session), (free_session),
(get_client), (free_client), (gst_rtp_bin_associate),
(free_stream), (gst_rtp_bin_class_init), (gst_rtp_bin_dispose),
(gst_rtp_bin_finalize):
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_rtp_jitter_buffer_finalize):
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_release):
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_finalize),
(gst_rtp_session_set_property), (gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_chain_send_rtp):
* gst/rtpmanager/gstrtpssrcdemux.c:
(gst_rtp_ssrc_demux_class_init), (gst_rtp_ssrc_demux_dispose):
* gst/rtpmanager/rtpsession.c: (rtp_session_finalize):
* gst/rtpmanager/rtpsession.h:
Various leak fixes.
2009-08-11 02:30:30 +01:00
Wim Taymans
919deb4490 gst/rtpmanager/gstrtpbin.c: Calculate and configure the NTP base time so that we can generate better
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (calc_ntp_ns_base),
(gst_rtp_bin_change_state), (new_payload_found), (create_send_rtp):
Calculate and configure the NTP base time so that we can generate better
NTP times in SR packets.
Set caps on new ghostpad.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_loop):
Clean debug statement.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_init), (gst_rtp_session_set_property),
(gst_rtp_session_get_property), (get_current_ntp_ns_time),
(rtcp_thread), (gst_rtp_session_event_recv_rtp_sink),
(gst_rtp_session_internal_links), (gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink),
(create_send_rtp_sink):
* gst/rtpmanager/gstrtpsession.h:
Add ntp-ns-base property to convert running_time to NTP time.
Handle NEWSEGMENT events on send and recv RTP pads so that we can
calculate the running time and thus NTP time of the packets.
Simplify getting the current NTP time using the pipeline clock.
Implement internal links functions.
Use the buffer timestamp to calculate the NTP time instead of the clock.
* gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc),
(gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_sink_event),
(gst_rtp_ssrc_demux_chain), (gst_rtp_ssrc_demux_rtcp_chain),
(gst_rtp_ssrc_demux_internal_links),
(gst_rtp_ssrc_demux_src_query):
* gst/rtpmanager/gstrtpssrcdemux.h:
Implement internal links function.
Calculate the diff between different streams, this might be used later
to get the inter stream latency.
* gst/rtpmanager/rtpsession.c: (rtp_session_send_rtp):
Simple cleanup.
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(calculate_jitter), (rtp_source_send_rtp), (rtp_source_get_new_sr):
Make the clock skew window a little bigger.
Apply the clock skew to all buffers, not just one with a new timestamp.
Calculate and debug sender clock drift.
Use extended last timestamp to interpollate for SR reports.
2009-08-11 02:30:29 +01:00
Tim-Philipp Müller
aa8985d1e4 gst/rtpmanager/gstrtpsession.c: Make compiler happy: fix compilation with -Wall -Werror (#473562).
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c:
Make compiler happy: fix compilation with -Wall -Werror
(#473562).
2009-08-11 02:30:29 +01:00
Wim Taymans
e7b6212c51 gst/rtpmanager/: Updated example pipelines in docs.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin-marshal.list:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_get_client),
(gst_rtp_bin_associate), (gst_rtp_bin_sync_chain), (create_stream),
(gst_rtp_bin_init), (caps_changed), (new_ssrc_pad_found),
(create_recv_rtp), (create_recv_rtcp), (create_send_rtp):
* gst/rtpmanager/gstrtpbin.h:
Updated example pipelines in docs.
Handle sync_rtcp buffers from the SSRC demuxer to perform lip-sync.
Set the default latency correctly.
Add some more points where we can get caps.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_query),
(gst_rtp_jitter_buffer_set_property),
(gst_rtp_jitter_buffer_get_property):
Add ts-offset property to control timestamping.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_init), (gst_rtp_session_set_property),
(gst_rtp_session_get_property), (get_current_ntp_ns_time),
(rtcp_thread), (stop_rtcp_thread), (gst_rtp_session_change_state),
(gst_rtp_session_send_rtcp), (gst_rtp_session_sync_rtcp),
(gst_rtp_session_cache_caps), (gst_rtp_session_clock_rate),
(gst_rtp_session_sink_setcaps), (gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink),
(create_recv_rtcp_sink), (create_send_rtp_sink),
(create_send_rtcp_src):
Various cleanups.
Feed rtpsession manager with NTP time based on pipeline clock when
handling RTP packets and RTCP timeouts.
Perform all RTCP with the system clock.
Set caps on RTCP outgoing buffers.
* gst/rtpmanager/gstrtpssrcdemux.c: (find_demux_pad_for_ssrc),
(create_demux_pad_for_ssrc), (gst_rtp_ssrc_demux_base_init),
(gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_sink_event),
(gst_rtp_ssrc_demux_rtcp_sink_event), (gst_rtp_ssrc_demux_chain),
(gst_rtp_ssrc_demux_rtcp_chain):
* gst/rtpmanager/gstrtpssrcdemux.h:
Also demux RTCP messages.
* gst/rtpmanager/rtpsession.c: (rtp_session_set_callbacks),
(update_arrival_stats), (rtp_session_process_rtp),
(rtp_session_process_rb), (rtp_session_process_sr),
(rtp_session_process_rr), (rtp_session_process_rtcp),
(rtp_session_send_rtp), (rtp_session_send_bye),
(session_start_rtcp), (session_report_blocks), (session_cleanup),
(rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Remove the get_time callback, the GStreamer part will feed us with
enough timing information.
Split sync timing and RTCP timing information.
Factor out common RB handling for SR and RR.
Send out SR RTCP packets for lip-sync.
Move SR and RR packet info generation to the source.
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(rtp_source_update_caps), (get_clock_rate), (calculate_jitter),
(rtp_source_process_rtp), (rtp_source_send_rtp),
(rtp_source_process_sr), (rtp_source_process_rb),
(rtp_source_get_new_sr), (rtp_source_get_new_rb),
(rtp_source_get_last_sr):
* gst/rtpmanager/rtpsource.h:
* gst/rtpmanager/rtpstats.h:
Use caps on incomming buffers to get timing information when they are
there.
Calculate clock scew of the receiver compared to the sender and adjust
the rtp timestamps.
Calculate the round trip in sources.
Do SR and RR calculations in the source.
2009-08-11 02:30:29 +01:00
Wim Taymans
f4e6f22315 gst/rtpmanager/gstrtpjitterbuffer.c: Use extended timestamp to release buffers from the jitterbuffer so that we can h...
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_flush_stop),
(gst_rtp_jitter_buffer_change_state), (gst_rtp_jitter_buffer_loop):
Use extended timestamp to release buffers from the jitterbuffer so that
we can handle the rtp wraparound correctly.
2009-08-11 02:30:29 +01:00
Wim Taymans
c576bcec15 gst/rtpmanager/gstrtpjitterbuffer.c: Improve Comments.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_loop):
Improve Comments.
* gst/rtpmanager/gstrtpsession.c: (stop_rtcp_thread),
(gst_rtp_session_change_state), (gst_rtp_session_parse_caps),
(gst_rtp_session_clock_rate), (gst_rtp_session_sink_setcaps),
(gst_rtp_session_event_send_rtp_sink), (create_recv_rtp_sink),
(create_send_rtp_sink):
Also parse the sink caps for clock-rate instead of only relying on the
result of the signal.
* gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp):
Make sure we fetch the clock rate for payloads we are sending out so
that we can use it for SR reports.
2009-08-11 02:30:29 +01:00
Wim Taymans
325dac0fc2 gst/rtpmanager/gstrtpsession.*: Distribute synchronisation parameters to the session manager so that it can generate ...
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (stop_rtcp_thread),
(gst_rtp_session_change_state),
(gst_rtp_session_event_send_rtp_sink):
* gst/rtpmanager/gstrtpsession.h:
Distribute synchronisation parameters to the session manager so that it
can generate correct SR packets for lip-sync.
* gst/rtpmanager/rtpsession.c: (rtp_session_set_base_time),
(rtp_session_set_timestamp_sync), (session_start_rtcp):
* gst/rtpmanager/rtpsession.h:
Add methods for setting sync parameters.
Set correct RTP time in SR packets using the sync params.
* gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp):
* gst/rtpmanager/rtpsource.h:
Record last RTP <-> GST timestamp so that we can use them to convert NTP
to RTP timestamps in SR packets.
2009-08-11 02:30:29 +01:00
Wim Taymans
eb86865a62 gst/rtpmanager/gstrtpbin.c: Add some more advanced example pipelines.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_clear_pt_map):
Add some more advanced example pipelines.
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
(stop_rtcp_thread), (gst_rtp_session_send_rtcp):
Add some debug and FIXME.
Release LOCK when performing session cleanup.
* gst/rtpmanager/rtpsession.c: (session_report_blocks):
Add some debug.
* gst/rtpmanager/rtpsource.c: (calculate_jitter),
(rtp_source_send_rtp):
Make sure we always send RTP packets with the session SSRC.
2009-08-11 02:30:29 +01:00
Wim Taymans
6835b966ec gst/rtpmanager/gstrtpjitterbuffer.c: When synchronizing buffers, take peer latency into account.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_change_state), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_query):
When synchronizing buffers, take peer latency into account.
Don't try to add our latency to invalid peer max latency values.
2009-08-11 02:30:29 +01:00
Tim-Philipp Müller
10d6ba4d61 Rename all GstRTPFoo structs to GstRtpFoo so that GST_BOILERPLATE registers a GType that's different than the GstRTPF...
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
* docs/plugins/gst-plugins-bad-plugins.interfaces:
* docs/plugins/gst-plugins-bad-plugins.signals:
* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpclient.h:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpptdemux.h:
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/gstrtpsession.h:
* gst/rtpmanager/gstrtpssrcdemux.c:
* gst/rtpmanager/gstrtpssrcdemux.h:
Rename all GstRTPFoo structs to GstRtpFoo so that GST_BOILERPLATE
registers a GType that's different than the GstRTPFoo types that
farsight registers (luckily GType names are case sensitive). Should
finally fix #430664.
2009-08-11 02:30:29 +01:00
Wim Taymans
f13ad91c77 gst/rtpmanager/gstrtpjitterbuffer.c: When drop-on-latency is set but we have no latency configured, just push the buf...
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_set_property):
When drop-on-latency is set but we have no latency configured, just push
the buffer as fast as possible.
Fix typo in comment.
2009-08-11 02:30:29 +01:00
Wim Taymans
ce70e0f43e gst/rtpmanager/rtpjitterbuffer.*: Fix undefined overflow prone ts_diff handling.
Original commit message from CVS:
* gst/rtpmanager/rtpjitterbuffer.c:
(rtp_jitter_buffer_get_ts_diff):
* gst/rtpmanager/rtpjitterbuffer.h:
Fix undefined overflow prone ts_diff handling.
2009-08-11 02:30:28 +01:00
Wim Taymans
076da98efb gst/rtpmanager/gstrtpjitterbuffer.c: Fix EOS handling.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_loop):
Fix EOS handling.
Convert some DEBUG into WARNINGs.
Pause task when flushing.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(rtcp_thread), (gst_rtp_session_event_recv_rtcp_sink):
Use system clock for RTCP session management timeouts.
* gst/rtpmanager/rtpsession.c: (on_new_ssrc), (on_ssrc_collision),
(on_ssrc_validated), (on_bye_ssrc), (on_bye_timeout), (on_timeout):
Release the session lock when emiting signals.
2009-08-11 02:30:28 +01:00
Stefan Kost
f24c54f4b5 gst/rtpmanager/rtpjitterbuffer.c: Include stdlib.
Original commit message from CVS:
* gst/rtpmanager/rtpjitterbuffer.c:
Include stdlib.
2009-08-11 02:30:28 +01:00
Wim Taymans
cdd82f2a95 gst/rtpmanager/: Remove complicated async queue and replace with more simple jitterbuffer code while also fixing some...
Original commit message from CVS:
* gst/rtpmanager/Makefile.am:
* gst/rtpmanager/async_jitter_queue.c:
* gst/rtpmanager/async_jitter_queue.h:
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_class_init),
(rtp_jitter_buffer_init), (rtp_jitter_buffer_finalize),
(rtp_jitter_buffer_new), (compare_seqnum),
(rtp_jitter_buffer_insert), (rtp_jitter_buffer_pop),
(rtp_jitter_buffer_flush), (rtp_jitter_buffer_num_packets),
(rtp_jitter_buffer_get_ts_diff):
* gst/rtpmanager/rtpjitterbuffer.h:
Remove complicated async queue and replace with more simple jitterbuffer
code while also fixing some bugs.
* gst/rtpmanager/gstrtpbin-marshal.list:
* gst/rtpmanager/gstrtpbin.c: (on_new_ssrc), (on_ssrc_collision),
(on_ssrc_validated), (on_bye_ssrc), (on_bye_timeout), (on_timeout),
(create_session), (gst_rtp_bin_class_init), (create_recv_rtp),
(create_send_rtp):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_dispose),
(gst_jitter_buffer_sink_parse_caps),
(gst_rtp_jitter_buffer_flush_start),
(gst_rtp_jitter_buffer_flush_stop),
(gst_rtp_jitter_buffer_change_state),
(gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_set_property):
* gst/rtpmanager/gstrtpsession.c: (on_new_ssrc),
(on_ssrc_collision), (on_ssrc_validated), (on_bye_ssrc),
(on_bye_timeout), (on_timeout), (gst_rtp_session_class_init),
(gst_rtp_session_init):
* gst/rtpmanager/gstrtpsession.h:
* gst/rtpmanager/rtpsession.c: (on_bye_ssrc), (session_cleanup):
Use new jitterbuffer code.
Expose some new signals in preparation for handling EOS.
2009-08-11 02:30:28 +01:00
Stefan Kost
366a756552 Add stdlib include (free, atoi, exit).
Original commit message from CVS:
* examples/app/appsrc_ex.c:
* examples/switch/switcher.c:
* ext/neon/gstneonhttpsrc.c:
* ext/timidity/gstwildmidi.c:
* ext/x264/gstx264enc.c:
* gst/mve/mveaudioenc.c: (mve_compress_audio):
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/spectrum/demo-audiotest.c:
* gst/spectrum/demo-osssrc.c:
* sys/dvb/gstdvbsrc.c:
Add stdlib include (free, atoi, exit).
2009-08-11 02:30:28 +01:00
Jens Granseuer
31571c8cb2 gst/: Build fixes for gcc-2.9x (no mid-block variable declarations etc.).
Original commit message from CVS:
Patch by: Jens Granseuer  <jensgr at gmx net>
* gst/equalizer/gstiirequalizer.c:
* gst/equalizer/gstiirequalizer10bands.c:
* gst/equalizer/gstiirequalizer3bands.c:
* gst/equalizer/gstiirequalizernbands.c:
* gst/rtpmanager/async_jitter_queue.c:
(async_jitter_queue_push_sorted):
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain):
* gst/switch/gstswitch.c: (gst_switch_chain):
Build fixes for gcc-2.9x (no mid-block variable declarations etc.).
Fixes #450185.
2009-08-11 02:30:28 +01:00
Wim Taymans
0c4fe985b6 Rename elements to avoid conflict with farsight elements with the same name. Fixes #430664.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* gst/rtpmanager/gstrtpbin.c: (create_session), (create_stream),
(gst_rtp_bin_class_init), (create_recv_rtp), (create_recv_rtcp),
(create_send_rtp), (create_rtcp), (gst_rtp_bin_request_new_pad):
* gst/rtpmanager/gstrtpclient.c: (create_stream),
(gst_rtp_client_request_new_pad):
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_clear_pt_map), (gst_rtp_jitter_buffer_loop):
* gst/rtpmanager/gstrtpmanager.c: (plugin_init):
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_request_new_pad):
* gst/rtpmanager/gstrtpssrcdemux.c:
Rename elements to avoid conflict with farsight elements with the same
name. Fixes #430664.
2009-08-11 02:30:28 +01:00
Wim Taymans
2a8cfc6410 Document stuff.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_rtp_jitter_buffer_clear_pt_map), (gst_rtp_jitter_buffer_loop):
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init),
(gst_rtp_pt_demux_clear_pt_map):
* gst/rtpmanager/gstrtpptdemux.h:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(rtcp_thread), (gst_rtp_session_clear_pt_map):
* gst/rtpmanager/gstrtpsession.h:
* gst/rtpmanager/gstrtpssrcdemux.c:
(gst_rtp_ssrc_demux_class_init):
Document stuff.
Add clear-pt-map action signal where needed.
2009-08-11 02:30:27 +01:00
Wim Taymans
3bc059707d gst/rtpmanager/gstrtpptdemux.c: We always use fixed caps.
Original commit message from CVS:
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
We always use fixed caps.
2009-08-11 02:30:27 +01:00
David Schleef
720dfeb3a5 gst/rtpmanager/gstrtpbin.c: g_hash_table_remove_all() only exists in 2.12. Work around.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c:
g_hash_table_remove_all() only exists in 2.12.  Work around.
2009-08-11 02:30:27 +01:00
Wim Taymans
62d401eb93 gst/rtpmanager/async_jitter_queue.c: Fix leak when flushing.
Original commit message from CVS:
* gst/rtpmanager/async_jitter_queue.c:
(async_jitter_queue_set_flushing_unlocked):
Fix leak when flushing.
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_clear_pt_map),
(gst_rtp_bin_class_init):
* gst/rtpmanager/gstrtpbin.h:
Add clear-pt-map signal.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_flush_stop),
(gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_loop):
Init clock-rate to -1 to mark unknow clock rate.
Fix flushing.
2009-08-11 02:30:27 +01:00
Stefan Kost
15b54ec7e2 gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream, gst_qtdemux_loop_state_header, gst_qtdemux_activate_segment, gst_qtde...
Original commit message from CVS:
* gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream,
gst_qtdemux_loop_state_header, gst_qtdemux_activate_segment,
gst_qtdemux_prepare_current_sample, gst_qtdemux_combine_flows,
gst_qtdemux_loop_state_movie, gst_qtdemux_loop,
qtdemux_parse_segments, qtdemux_parse_trak):
* gst/rtpmanager/rtpsession.c (rtp_session_get_bandwidth,
rtp_session_get_rtcp_bandwidth, rtp_session_get_cname,
rtp_session_get_name, rtp_session_get_email, rtp_session_get_phone,
rtp_session_get_location, rtp_session_get_tool,
rtp_session_process_bye, session_report_blocks):
* gst/rtpmanager/rtpsource.c (rtp_source_process_rtp,
rtp_source_send_rtp, rtp_source_process_sr, rtp_source_process_rb):
More format arg fixing (spotted by Ali Sabil <ali.sabil@gmail.com>).
* gst/switch/Makefile.am:
Add require libraries(spotted by Ali Sabil <ali.sabil@gmail.com>).
2009-08-11 02:30:27 +01:00
Stefan Kost
091c2cfbc0 gst/rtpmanager/async_jitter_queue.c (tail_buffer_duration, async_jitter_queue_ref, async_jitter_queue_ref_unlocked, a...
Original commit message from CVS:
* gst/rtpmanager/async_jitter_queue.c (tail_buffer_duration,
async_jitter_queue_ref, async_jitter_queue_ref_unlocked,
async_jitter_queue_set_low_threshold,
async_jitter_queue_length_ts_units_unlocked,
async_jitter_queue_unref_and_unlock, async_jitter_queue_unref,
async_jitter_queue_lock, async_jitter_queue_push,
async_jitter_queue_push_unlocked, async_jitter_queue_push_sorted,
async_jitter_queue_pop_intern_unlocked, async_jitter_queue_pop,
async_jitter_queue_pop_unlocked, async_jitter_queue_length_unlocked,
async_jitter_queue_set_flushing_unlocked,
async_jitter_queue_unset_flushing_unlocked):
Format arg fix (spotted by Ali Sabil <ali.sabil@gmail.com>)
2009-08-11 02:30:27 +01:00
Wim Taymans
88f2441722 gst/rtpmanager/gstrtpjitterbuffer.c: Pass queries upstream.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_query):
Pass queries upstream.
2009-08-11 02:30:27 +01:00
Wim Taymans
a241c62ecb gst/rtpmanager/gstrtpjitterbuffer.c: Add some debug info.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_query):
Add some debug info.
* gst/rtpmanager/rtpsession.c: (rtp_session_init),
(rtp_session_send_rtp):
Store real user name in the session.
2009-08-11 02:30:27 +01:00
Wim Taymans
600afaaff9 gst/rtpmanager/async_jitter_queue.c: Fix the case where the buffer underruns and does not block.
Original commit message from CVS:
* gst/rtpmanager/async_jitter_queue.c: (signal_waiting_threads),
(async_jitter_queue_pop_intern_unlocked):
Fix the case where the buffer underruns and does not block.
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_base_init),
(create_recv_rtcp), (create_send_rtp), (create_rtcp),
(gst_rtp_bin_request_new_pad):
Rename RTCP send pad, like in the session manager.
Allow getting an RTCP pad for receiving even if we don't receive RTP.
fix handling of send_rtp_src pad.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
When no pt map could be found, fall back to the sinkpad caps.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
(gst_rtp_session_send_rtp), (create_recv_rtp_sink),
(create_recv_rtcp_sink), (create_send_rtp_sink),
(create_send_rtcp_src):
Fix pad names.
* gst/rtpmanager/rtpsession.c: (source_push_rtp),
(rtp_session_create_source), (rtp_session_process_sr),
(rtp_session_send_rtp), (session_start_rtcp):
* gst/rtpmanager/rtpsession.h:
Unlock session when performing a callback.
Add callbacks for the internal session object.
Fix sending of RTP packets.
first attempt at adding NTP times in the SR packets.
Small debug and doc improvements.
* gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp):
Update stats for SR reports.
2009-08-11 02:30:26 +01:00
Wim Taymans
e6537bcd7c gst/rtpmanager/gstrtpsession.c: Remove debug.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_send_rtcp):
Remove debug.
* gst/rtpmanager/rtpsession.c: (rtp_session_process_sr),
(rtp_session_process_sdes), (calculate_rtcp_interval),
(rtp_session_next_timeout), (session_report_blocks):
* gst/rtpmanager/rtpstats.c: (rtp_stats_calculate_rtcp_interval):
Improve debugging
Fix interval for BYE/RTCP packets.
2009-08-11 02:30:26 +01:00
Wim Taymans
a7b80281d1 gst/rtpmanager/gstrtpsession.c: Move reconsideration code to the rtpsession object.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
(gst_rtp_session_send_rtcp), (gst_rtp_session_reconsider):
Move reconsideration code to the rtpsession object.
Simplify timout handling and add reconsideration.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (rtp_session_finalize), (on_bye_ssrc),
(on_bye_timeout), (on_timeout), (rtp_session_set_callbacks),
(obtain_source), (rtp_session_create_source),
(update_arrival_stats), (rtp_session_process_rtp),
(rtp_session_process_sr), (rtp_session_process_rr),
(rtp_session_process_bye), (rtp_session_process_rtcp),
(calculate_rtcp_interval), (rtp_session_send_bye),
(rtp_session_next_timeout), (session_start_rtcp),
(session_report_blocks), (session_cleanup), (session_sdes),
(session_bye), (is_rtcp_time), (rtp_session_on_timeout):
* gst/rtpmanager/rtpsession.h:
Handle timeout of inactive sources and senders.
Implement BYE scheduling.
* gst/rtpmanager/rtpsource.c: (calculate_jitter),
(rtp_source_process_sr), (rtp_source_get_last_sr),
(rtp_source_get_last_rb):
* gst/rtpmanager/rtpsource.h:
Add members to check for timeouts.
* gst/rtpmanager/rtpstats.c: (rtp_stats_init_defaults),
(rtp_stats_calculate_rtcp_interval), (rtp_stats_add_rtcp_jitter),
(rtp_stats_calculate_bye_interval):
* gst/rtpmanager/rtpstats.h:
Use RFC algorithm for calculating the reporting interval.
2009-08-11 02:30:26 +01:00
Wim Taymans
43f0b878c9 gst/rtpmanager/gstrtpsession.c: Implement forward and reverse reconsideration.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread):
Implement forward and reverse reconsideration.
* gst/rtpmanager/rtpsession.c: (rtp_session_get_num_sources),
(rtp_session_get_num_active_sources), (rtp_session_process_sr),
(session_report_blocks):
* gst/rtpmanager/rtpsession.h:
Small cleanups.
2009-08-11 02:30:26 +01:00
Wim Taymans
333764307d gst/rtpmanager/gstrtpbin.*: Make default jitterbuffer latency configurable.
Original commit message from CVS:
reviewed by: <delete if not using a buddy>
* gst/rtpmanager/gstrtpbin.c: (create_stream),
(gst_rtp_bin_class_init), (gst_rtp_bin_set_property),
(gst_rtp_bin_get_property):
* gst/rtpmanager/gstrtpbin.h:
Make default jitterbuffer latency configurable.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_set_property),
(gst_rtp_jitter_buffer_get_property):
Debuging cleanups.
2009-08-11 02:30:26 +01:00
Wim Taymans
ae536e0c89 gst/rtpmanager/gstrtpjitterbuffer.c: Report NO_PREROLL when going to PAUSED.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_change_state):
Report NO_PREROLL when going to PAUSED.
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread):
Don't send RTCP right before we are shutting down.
* gst/rtpmanager/rtpsession.c: (rtp_session_process_rtp),
(rtp_session_process_sr), (session_report_blocks),
(rtp_session_perform_reporting):
Improve report blocks.
* gst/rtpmanager/rtpsource.c: (calculate_jitter), (init_seq),
(rtp_source_process_rtp), (rtp_source_process_sr),
(rtp_source_process_rb), (rtp_source_get_last_sr),
(rtp_source_get_last_rb):
* gst/rtpmanager/rtpsource.h:
* gst/rtpmanager/rtpstats.h:
Cleanups, add methods to access stats.
2009-08-11 02:30:25 +01:00
Wim Taymans
23883be047 gst/rtpmanager/gstrtpbin.c: fix for pad name change
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_rtcp):
fix for pad name change
* gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
(gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate):
Fix for renamed methods.
* gst/rtpmanager/rtpsession.c: (rtp_session_init),
(rtp_session_finalize), (rtp_session_set_cname),
(rtp_session_get_cname), (rtp_session_set_name),
(rtp_session_get_name), (rtp_session_set_email),
(rtp_session_get_email), (rtp_session_set_phone),
(rtp_session_get_phone), (rtp_session_set_location),
(rtp_session_get_location), (rtp_session_set_tool),
(rtp_session_get_tool), (rtp_session_set_note),
(rtp_session_get_note), (source_push_rtp), (obtain_source),
(rtp_session_add_source), (rtp_session_get_source_by_ssrc),
(rtp_session_create_source), (rtp_session_process_rtp),
(rtp_session_process_sr), (rtp_session_process_sdes),
(rtp_session_process_rtcp), (rtp_session_send_rtp),
(rtp_session_get_reporting_interval), (session_report_blocks),
(session_sdes), (rtp_session_perform_reporting):
* gst/rtpmanager/rtpsession.h:
Prepare for implementing SSRC sampling.
Create SSRC for the session.
Add methods to set the SDES entries.
fix accounting of senders/receivers.
Implement SR/RR/SDES RTCP reporting.
* gst/rtpmanager/rtpsource.c: (rtp_source_init), (init_seq),
(rtp_source_process_rtp), (rtp_source_process_sr):
* gst/rtpmanager/rtpsource.h:
Implement extended sequence number.
* gst/rtpmanager/rtpstats.c: (rtp_stats_calculate_rtcp_interval):
* gst/rtpmanager/rtpstats.h:
Rename some fields.
2009-08-11 02:30:25 +01:00
Tim-Philipp Müller
677b361dc3 gst/rtpmanager/rtpsession.c: Don't use GLib-2.10 API, we only require GLib 2.8 at the moment.
Original commit message from CVS:
* gst/rtpmanager/rtpsession.c: (rtp_session_finalize):
Don't use GLib-2.10 API, we only require GLib 2.8 at the moment.
2009-08-11 02:30:25 +01:00
Wim Taymans
54b3dec1f5 configure.ac: Disable rtpmanager for now because it depends on CVS -base.
Original commit message from CVS:
* configure.ac:
Disable rtpmanager for now because it depends on CVS -base.
* gst/rtpmanager/Makefile.am:
Added new files for session manager.
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
(create_stream), (pt_map_requested), (new_ssrc_pad_found):
Some cleanups.
the session manager can now also request a pt-map.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init),
(gst_rtp_session_class_init), (gst_rtp_session_init),
(gst_rtp_session_finalize), (rtcp_thread), (start_rtcp_thread),
(stop_rtcp_thread), (gst_rtp_session_change_state),
(gst_rtp_session_process_rtp), (gst_rtp_session_send_rtp),
(gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate),
(gst_rtp_session_get_time), (gst_rtp_session_event_recv_rtp_sink),
(gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_event_recv_rtcp_sink),
(gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp), (create_send_rtcp_src),
(gst_rtp_session_request_new_pad):
* gst/rtpmanager/gstrtpsession.h:
We can ask for pt-map now too when the session manager needs it.
Hook up to the new session manager, implement the needed callbacks for
pushing data, getting clock time and requesting clock-rates.
Rename rtcp_src to send_rtcp_src to make it clear that this RTCP is to
be send to clients.
Add code to start and stop the thread that will schedule RTCP through
the session manager.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (rtp_session_finalize),
(rtp_session_set_property), (rtp_session_get_property),
(on_new_ssrc), (on_ssrc_collision), (on_ssrc_validated),
(on_bye_ssrc), (rtp_session_new), (rtp_session_set_callbacks),
(rtp_session_set_bandwidth), (rtp_session_get_bandwidth),
(rtp_session_set_rtcp_bandwidth), (rtp_session_get_rtcp_bandwidth),
(source_push_rtp), (source_clock_rate), (check_collision),
(obtain_source), (rtp_session_add_source),
(rtp_session_get_num_sources),
(rtp_session_get_num_active_sources),
(rtp_session_get_source_by_ssrc),
(rtp_session_get_source_by_cname), (rtp_session_create_source),
(update_arrival_stats), (rtp_session_process_rtp),
(rtp_session_process_sr), (rtp_session_process_rr),
(rtp_session_process_sdes), (rtp_session_process_bye),
(rtp_session_process_app), (rtp_session_process_rtcp),
(rtp_session_send_rtp), (rtp_session_get_rtcp_interval),
(rtp_session_produce_rtcp):
* gst/rtpmanager/rtpsession.h:
The advanced beginnings of the main session manager that handles the
participant database of RTPSources, SSRC probation, SSRC collisions,
parse RTCP to update source stats. etc..
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_init), (rtp_source_finalize), (rtp_source_new),
(rtp_source_set_callbacks), (rtp_source_set_as_csrc),
(rtp_source_set_rtp_from), (rtp_source_set_rtcp_from),
(push_packet), (get_clock_rate), (calculate_jitter),
(rtp_source_process_rtp), (rtp_source_process_bye),
(rtp_source_send_rtp), (rtp_source_process_sr),
(rtp_source_process_rb):
* gst/rtpmanager/rtpsource.h:
Object that encapsulates an SSRC and its state in the database.
Calculates the jitter and transit times of data packets.
* gst/rtpmanager/rtpstats.c: (rtp_stats_init_defaults),
(rtp_stats_calculate_rtcp_interval), (rtp_stats_add_rtcp_jitter):
* gst/rtpmanager/rtpstats.h:
Various stats regarding the session and sources.
Used to calculate the RTCP interval.
2009-08-11 02:30:25 +01:00
Wim Taymans
490113d40d gst/rtpmanager/: Protect lists and structures with locks.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
(gst_rtp_bin_init), (gst_rtp_bin_finalize), (new_ssrc_pad_found),
(create_recv_rtp), (gst_rtp_bin_request_new_pad):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_init), (gst_rtp_session_finalize),
(gst_rtp_session_event_recv_rtp_sink),
(gst_rtp_session_event_recv_rtcp_sink),
(gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_request_new_pad):
Protect lists and structures with locks.
Return FLOW_OK from RTCP messages for now.
2009-08-11 02:30:25 +01:00
Wim Taymans
8bbea77a41 gst/rtpmanager/gstrtpbin.c: Emit pt map requests and cache results.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
(create_stream), (gst_rtp_bin_class_init), (pt_map_requested):
Emit pt map requests and cache results.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_jitter_buffer_sink_parse_caps),
(gst_jitter_buffer_sink_setcaps),
(gst_rtp_jitter_buffer_get_clock_rate),
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
Emit request-pt-map signals.
2009-08-11 02:30:25 +01:00
Wim Taymans
03bf43d50e gst/rtpmanager/gstrtpbin-marshal.list: Some more custom marshallers.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin-marshal.list:
Some more custom marshallers.
* gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
(clock_rate_request), (create_stream), (gst_rtp_bin_class_init),
(pt_map_requested), (new_ssrc_pad_found), (create_recv_rtp):
* gst/rtpmanager/gstrtpbin.h:
Prepare for caching pt maps.
Connect to signals to collect pt maps.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init),
(gst_jitter_buffer_sink_setcaps), (gst_rtp_jitter_buffer_loop):
* gst/rtpmanager/gstrtpjitterbuffer.h:
Add request_clock_rate signal.
Use scale insteat of scale_int because the later does not deal with
negative numbers.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init),
(gst_rtp_pt_demux_chain):
* gst/rtpmanager/gstrtpptdemux.h:
Implement request-pt-map signal.
2009-08-11 02:30:24 +01:00
Wim Taymans
8c67b5d7dd gst/rtpmanager/: Added custom marshallers for signals.
Original commit message from CVS:
* gst/rtpmanager/.cvsignore:
* gst/rtpmanager/Makefile.am:
* gst/rtpmanager/gstrtpbin-marshal.list:
Added custom marshallers for signals.
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init):
* gst/rtpmanager/gstrtpbin.h:
Prepare for emiting pt map signals.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init):
* gst/rtpmanager/gstrtpssrcdemux.c:
(gst_rtp_ssrc_demux_class_init):
Fix signals.
2009-08-11 02:30:24 +01:00
Wim Taymans
a6aa41dc21 gst/rtpmanager/gstrtpbin.*: Provide a clock.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init),
(gst_rtp_bin_init), (gst_rtp_bin_provide_clock):
* gst/rtpmanager/gstrtpbin.h:
Provide a clock.
2009-08-11 02:30:24 +01:00
Wim Taymans
1b0ae2608f gst/rtpmanager/gstrtpbin.c: Fix pad template name parsing.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_rtcp):
Fix pad template name parsing.
2009-08-11 02:30:24 +01:00
Wim Taymans
63dbc75734 gst/rtpmanager/gstrtpjitterbuffer.c: Add some debug and comments.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_jitter_buffer_sink_setcaps), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_loop):
Add some debug and comments.
Fix double unref() in error cases.
2009-08-11 02:30:24 +01:00
Wim Taymans
9bfc641f0d gst/rtpmanager/gstrtpbin.*: Add debugging category.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (find_session_by_id),
(create_session), (find_stream_by_ssrc), (create_stream),
(gst_rtp_bin_class_init), (new_payload_found),
(new_ssrc_pad_found), (create_recv_rtp), (create_recv_rtcp),
(create_send_rtp), (create_rtcp):
* gst/rtpmanager/gstrtpbin.h:
Add debugging category.
Added RTPStream to manage stream per SSRC, each with its own
jitterbuffer and ptdemux.
Added SSRCDemux.
Connect to various SSRC and PT signals and create ghostpads, link stuff.
* gst/rtpmanager/gstrtpmanager.c: (plugin_init):
Added rtpbin to elements.
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
Fix caps and forward GstFlowReturn
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(gst_rtp_session_event_recv_rtp_sink),
(gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_event_recv_rtcp_sink),
(gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink),
(create_recv_rtcp_sink), (create_send_rtp_sink), (create_rtcp_src),
(gst_rtp_session_request_new_pad):
Add debug category.
Add event handling
* gst/rtpmanager/gstrtpssrcdemux.c: (find_rtp_pad_for_ssrc),
(create_rtp_pad_for_ssrc), (gst_rtp_ssrc_demux_class_init),
(gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_chain),
(gst_rtp_ssrc_demux_change_state):
* gst/rtpmanager/gstrtpssrcdemux.h:
Add debug category.
Add new-pt-pad signal.
2009-08-11 02:30:24 +01:00
Wim Taymans
a9d14ed310 gst/rtpmanager/: Added simple SSRC demuxer.
Original commit message from CVS:
* gst/rtpmanager/Makefile.am:
* gst/rtpmanager/gstrtpmanager.c: (plugin_init):
* gst/rtpmanager/gstrtpssrcdemux.c: (find_pad_for_ssrc),
(create_pad_for_ssrc), (gst_rtp_ssrc_demux_base_init),
(gst_rtp_ssrc_demux_class_init), (gst_rtp_ssrc_demux_init),
(gst_rtp_ssrc_demux_finalize), (gst_rtp_ssrc_demux_sink_event),
(gst_rtp_ssrc_demux_chain), (gst_rtp_ssrc_demux_src_event),
(gst_rtp_ssrc_demux_change_state):
* gst/rtpmanager/gstrtpssrcdemux.h:
Added simple SSRC demuxer.
2009-08-11 02:30:23 +01:00
Wim Taymans
5351f0cb51 gst/rtpmanager/: Some more ghostpad magic.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (find_session_by_id),
(create_session), (gst_rtp_bin_base_init), (create_recv_rtp),
(create_recv_rtcp), (create_send_rtp), (create_rtcp),
(gst_rtp_bin_request_new_pad):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpclient.c:
Some more ghostpad magic.
2009-08-11 02:30:23 +01:00
Wim Taymans
fdae491de7 gst/rtpmanager/Makefile.am: Add .h file so it can be disted properly.
Original commit message from CVS:
* gst/rtpmanager/Makefile.am:
Add .h file so it can be disted properly.
2009-08-11 02:30:23 +01:00
Wim Taymans
f0d1ab1c1f Add RTP session management elements. Still in progress.
Original commit message from CVS:
* configure.ac:
* gst/rtpmanager/Makefile.am:
* gst/rtpmanager/async_jitter_queue.c: (async_jitter_queue_new),
(signal_waiting_threads), (async_jitter_queue_ref),
(async_jitter_queue_ref_unlocked),
(async_jitter_queue_set_low_threshold),
(async_jitter_queue_set_high_threshold),
(async_jitter_queue_set_max_queue_length),
(async_jitter_queue_get_g_queue), (calculate_ts_diff),
(async_jitter_queue_length_ts_units_unlocked),
(async_jitter_queue_unref_and_unlock), (async_jitter_queue_unref),
(async_jitter_queue_lock), (async_jitter_queue_unlock),
(async_jitter_queue_push), (async_jitter_queue_push_unlocked),
(async_jitter_queue_push_sorted),
(async_jitter_queue_push_sorted_unlocked),
(async_jitter_queue_insert_after_unlocked),
(async_jitter_queue_pop_intern_unlocked), (async_jitter_queue_pop),
(async_jitter_queue_pop_unlocked), (async_jitter_queue_length),
(async_jitter_queue_length_unlocked),
(async_jitter_queue_set_flushing_unlocked),
(async_jitter_queue_unset_flushing_unlocked),
(async_jitter_queue_set_blocking_unlocked):
* gst/rtpmanager/async_jitter_queue.h:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_base_init),
(gst_rtp_bin_class_init), (gst_rtp_bin_init),
(gst_rtp_bin_finalize), (gst_rtp_bin_set_property),
(gst_rtp_bin_get_property), (gst_rtp_bin_change_state),
(gst_rtp_bin_request_new_pad), (gst_rtp_bin_release_pad):
* gst/rtpmanager/gstrtpbin.h:
* gst/rtpmanager/gstrtpclient.c: (new_pad), (create_stream),
(free_stream), (find_stream_by_ssrc), (gst_rtp_client_base_init),
(gst_rtp_client_class_init), (gst_rtp_client_init),
(gst_rtp_client_finalize), (gst_rtp_client_set_property),
(gst_rtp_client_get_property), (gst_rtp_client_change_state),
(gst_rtp_client_request_new_pad), (gst_rtp_client_release_pad):
* gst/rtpmanager/gstrtpclient.h:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_base_init),
(gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_init),
(gst_rtp_jitter_buffer_dispose), (gst_rtp_jitter_buffer_getcaps),
(gst_jitter_buffer_sink_setcaps), (free_func),
(gst_rtp_jitter_buffer_flush_start),
(gst_rtp_jitter_buffer_flush_stop),
(gst_rtp_jitter_buffer_src_activate_push),
(gst_rtp_jitter_buffer_change_state), (priv_compare_rtp_seq_lt),
(compare_rtp_buffers_seq_num), (gst_rtp_jitter_buffer_sink_event),
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_query),
(gst_rtp_jitter_buffer_set_property),
(gst_rtp_jitter_buffer_get_property):
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpmanager.c: (plugin_init):
* gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_base_init),
(gst_rtp_pt_demux_class_init), (gst_rtp_pt_demux_init),
(gst_rtp_pt_demux_finalize), (gst_rtp_pt_demux_chain),
(gst_rtp_pt_demux_getcaps), (find_pad_for_pt),
(gst_rtp_pt_demux_setup), (gst_rtp_pt_demux_release),
(gst_rtp_pt_demux_change_state):
* gst/rtpmanager/gstrtpptdemux.h:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init),
(gst_rtp_session_class_init), (gst_rtp_session_init),
(gst_rtp_session_finalize), (gst_rtp_session_set_property),
(gst_rtp_session_get_property), (gst_rtp_session_change_state),
(gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink),
(create_recv_rtcp_sink), (create_send_rtp_sink), (create_rtcp_src),
(gst_rtp_session_request_new_pad), (gst_rtp_session_release_pad):
* gst/rtpmanager/gstrtpsession.h:
Add RTP session management elements. Still in progress.
2009-08-11 02:30:23 +01:00
Mark Nauwelaerts
96e72522fc avidemux: push mode; cater for chunk padding 2009-08-10 14:41:52 +02:00
Mark Nauwelaerts
f67db2a089 avidemux: only use stream's pad after having checked it exists 2009-08-10 14:41:34 +02:00
Mark Nauwelaerts
4249f52c6c avidemux: sprinkle some more GST_DEBUG_FUNCPTR 2009-08-10 14:41:29 +02:00
Mark Nauwelaerts
6d26594eef avidemux: post error message if no pads to push EOS event on 2009-08-10 14:41:27 +02:00
Mark Nauwelaerts
b0a0c06155 avidemux: fix typo in warning message 2009-08-10 14:41:23 +02:00
Mark Nauwelaerts
7750173244 avidemux: fix some buffer ref handling 2009-08-10 14:41:19 +02:00
Mark Nauwelaerts
5b0f7f04e7 avidemux: do not exceed maximum number of supported streams 2009-08-10 14:41:16 +02:00
Mark Nauwelaerts
effa7b4660 avidemux: prevent double unref; gst_avi_demux_parse_avih already unrefs 2009-08-10 14:41:14 +02:00
Mark Nauwelaerts
42bc085d95 avidemux: verify size of INFO LIST to satisfy subsequent expectations 2009-08-10 14:41:12 +02:00
Mark Nauwelaerts
f4f8e8532c avidemux: check video stream framerate against avi header frame duration
The former might be bogus in silly cases, and the latter seems to
carry more weight.
2009-08-10 14:41:09 +02:00