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* rtph263pay: Default to doing A, B and C modes, not only AOlivier Crête2009-06-101-1/+1
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* deinterlace: Fix QoS calculationsSebastian Dröge2009-06-101-3/+6
| | | | | | | The diff is a signed integer, not an unsigned one of course. In modes other than GST_DEINTERLACE_ALL every frame has twice the duration of the field duration.
* rtpsirenpay: Put the bitrate in the RTP capsOlivier Crête2009-06-091-1/+2
| | | | | The MS code seems to require the bitrate to interoperate and draft-ietf-avt-rtp-g7221-00 also has it.
* deinterlace: Implement basic QoSSebastian Dröge2009-06-092-19/+158
| | | | | This change is based on Tim's QoS implementation for jpegdec.
* deinterlace: Directly proxy events/queries to the peer padsSebastian Dröge2009-06-091-6/+43
| | | | | This removes some overhead introduced by the default handlers that need to iterate over the other pads.
* avidemux: debug_memdump() unknown tags. Refactor junk parsing code.Edward Hervey2009-06-091-28/+47
| | | | This makes life slightly easier when debugging avi files.
* rtp: Don't forget to dist the headers for the CELT (de)payloaders.Edward Hervey2009-06-081-0/+2
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* Revert "Revert "qtdemux: fill timestamp table completely""Wim Taymans2009-06-071-3/+18
| | | | | | This reverts commit 9f022c8a8503c2ce0fa617fdb50e41706dd412f5. Sorry, I was thinking about the wrong module.
* Revert "qtdemux: fill timestamp table completely"Wim Taymans2009-06-071-18/+3
| | | | | This reverts commit 790b050fc5302cae89cddcd23b258093967d05a9. I forgot we were frozen.
* qtdemux: fill timestamp table completelyWim Taymans2009-06-071-3/+18
| | | | | | | | | | When there are less timestamps that there are samples, fill up the sample table with the last know timestamp. This situation can happen when the last sample does not decode and doesn't need a timestamp. We however calculate the total track length using the last sample timestamp so we need to have something sensible in there. Fixes #585056
* wavparse: handle LIST INFO of 0 sizeWim Taymans2009-06-071-21/+28
| | | | | | Handle LIST INFO chunks of 0 size instead of causing errors. Fixes #584981
* Revert "wavparse: Remove dead assignments, move variable to where it's needed."Wim Taymans2009-06-071-11/+16
| | | | | Reverts commit 44256a78f8dd79a91f3bb2ab7c3aa623c097bb8a and use the result in error reporting so that we can see what's going on.
* celtdepay: add CELT depayloaderWim Taymans2009-06-054-0/+298
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* rtpceltpay: add CELT RTP payloaderWim Taymans2009-06-054-0/+519
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* sunaudio: Fix switch setting on some devices. Add debug. Fix a FIXME.Jan Schmidt2009-06-053-21/+42
| | | | | | | | | | Fix the setting of toggle switches on some broken audio drivers which report that no audio ports are settable by ignoring the mod_port field there. Add some debug statements. Fix a FIXME now that Good relies on a new enough gst-plugins-base.
* sunaudio: Support new flags for options and actionsJan Schmidt2009-06-057-111/+583
| | | | | | | | | Use new audio mixer flags added in Base 0.10.23 to expose flags and options on the SunAudio devices. Fixes: #583593 Patch By: Brian Cameron <brian.cameron@sun.com> Patch By: Garrett D'Amore <garrett.damore@sun.com>
* deinterlace: First try to handle DVD still frames correctlySebastian Dröge2009-06-042-3/+63
| | | | This helps a bit with bug #582740 but still doesn't make it work.
* pulsesink: only notify if all checks passedStefan Kost2009-06-041-4/+3
| | | | | Replace goto done: with return, as those are checks when we don't want to flag a pending notify.
* rtspsrc: set the right state on rtpbinWim Taymans2009-06-041-1/+7
| | | | | We need to set the state of gstrtpbin to the same state as our source elements. This fixes fallback to TCP again.
* pulsesink: check pointer before accessingStefan Kost2009-06-031-3/+3
| | | | Move existing check a few lines up, so that we check before accessing fields.
* pulsesink: rename gst_pulse_sink_get_time to gst_pulsesink_get_timeStefan Kost2009-06-031-2/+2
| | | | Rename internal method for consistency.
* pulsesink: use values from pa_stream_get_buffer_attr()Stefan Kost2009-06-031-4/+6
| | | | | We were putting the requested values back into ringbuffer spec, instead of using the queried values.
* vrawpay: trim output buffersWim Taymans2009-06-021-0/+4
| | | | | Remove the leftover unused bytes in the output buffer. Fixes #584613
* vrawdepay: fix parsing of sampling fieldWim Taymans2009-06-021-1/+1
| | | | | commit a12d9a80f225be97b3674b1a0506ac66544dbf49 broke the parsing of the sampling.
* pngdec: Avoid possible overflow in calculationsJan Schmidt2009-06-021-4/+16
| | | | | | A malformed (or simply huge) PNG file can lead to integer overflow in calculating the size of the output buffer, leading to crashes or buffer overflows later. Fixes SA35205 security advisory.
* flacenc: some more logging - dump header packetsTim-Philipp Müller2009-06-021-6/+12
| | | | | Also, the final fixing up of the headers is expected and not something we should warn about.
* flacenc: never ever pass values >36bits to _set_total_samples_estimate()Tim-Philipp Müller2009-06-021-1/+1
| | | | | | | | | Let's be paranoid and make sure we never pass a number that takes up more than 36 bits to _set_total_samples_estimate(), since libFLAC expects all the other bits to be zero, and if this is not the case neighbouring fields in the global stream info header may get messed up inadvertently, so that flac -d refuses to decode the stream. See #584455.
* Address bad FLAC sample length encoding of #5844455Thomas Vander Stichele2009-06-011-1/+5
| | | | | | Commit df707c666433a78d3878af6f055698d5756226c4 introduced an obvious bug in the sample length calculation, using the wrong macro for conversion.
* deinterlace: Fix spurious colons in asm codeBrian Cameron2009-06-011-13/+13
| | | | | | Fixes #584174. Signed-off-by: David Schleef <ds@schleef.org>
* avidemux: skip JUNK chunks in data section in streaming modeTim-Philipp Müller2009-06-011-0/+2
| | | | | Skip JUNK tags in streaming mode as well instead of EOSing prematurely. Fixes #564100.
* videomixer: Don't use // commentsSebastian Dröge2009-05-283-6/+5
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* videomixer: Fix background blitting when a color mode is selected with BGRASebastian Dröge2009-05-281-7/+8
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* videomixer: Some cleanup and fix the calculation of the frame size in bytesSebastian Dröge2009-05-285-30/+44
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* videomixer: Fix I420 blending to actually do somethingSebastian Dröge2009-05-281-10/+33
| | | | | | For this we a) implement the checkers filling and b) actually blend the src/dest by using the src alpha value from the pad.
* videomixer: Fix ARGB blending to actually workSebastian Dröge2009-05-281-1/+1
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* videomixer: Blend BGRA ourselves instead of using CairoSebastian Dröge2009-05-282-28/+61
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* videomixer: Add support for blending BGRA and AYUVAlex Ugarte2009-05-286-283/+878
| | | | Fixes bug #577017.
* equalizer: Use floating point arithmetic internally for the int16 modeGhislain 'Aus' Lacroix2009-05-281-6/+6
| | | | | By using int32 arithmetic we will introduce distortions as the IIR filter is very sensitive to rounding errors. Fixes bug #580214.
* Update spec file with latest pluginsChristian Schaller2009-05-281-0/+3
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* Automatic update of common submoduleJan Schmidt2009-05-261-0/+0
| | | | From 888e0a2 to c572721
* v4l2: cleanup and commentingStefan Kost2009-05-262-48/+17
| | | | | Remove newlines inserted by gst-indent once. Remove unused var from instance struct. Add comments. Add another #define for default property value.
* makefile: idea about makeing more sources/sinks testable againStefan Kost2009-05-261-1/+14
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* pngdec: match g_malloc() with g_free()John Keeping2009-05-251-1/+1
| | | | | | Matching g_malloc() with a g_free() is important when a custom allocator is installed. Fixes #583803
* rtpmp4vpay: don't look for headers in some casesWim Taymans2009-05-252-1/+7
| | | | | | In some streams (starting with 00000100) don't look for the headers but push data as it is. Fixes #582153
* rtspsrc: fix memory leak of messagesPatrick Radizi2009-05-251-1/+6
| | | | | Free messages correctly. Fixes #577318
* rtspsrc: make fakesrc silentWim Taymans2009-05-241-1/+1
| | | | Make the fakesrc that is responsible for sending dummy packets silent.
* rtspsrc: don't send teardown before setupWim Taymans2009-05-241-0/+4
| | | | | Don't send a TEARDOWN request when we did not manage to successfully setup a stream.
* matroskademux: Populate a GstIndex that is set on matroskademuxSebastian Dröge2009-05-223-3/+111
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* flvmux: Get the max duration from upstream if there's no duration tagSebastian Dröge2009-05-221-14/+40
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* flvmux: Write an index table to the end of the fileSebastian Dröge2009-05-222-5/+151
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