Commit graph

5067 commits

Author SHA1 Message Date
Sebastian Dröge
b641c49a4c tests/check/elements/deinterleave.c: Set keep-positions property to TRUE for the 8 channel test to ensure that the or...
Original commit message from CVS:
* tests/check/elements/deinterleave.c: (GST_START_TEST):
Set keep-positions property to TRUE for the 8 channel test to ensure
that the original channel position is set on the output.
2008-05-19 12:32:06 +00:00
Sebastian Dröge
814b6ea02c gst/interleave/deinterleave.*: Add a property to select whether channel positions should be kept on the mono output b...
Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_class_init),
(gst_deinterleave_init), (gst_deinterleave_add_new_pads),
(gst_deinterleave_set_pads_caps), (gst_deinterleave_set_property),
(gst_deinterleave_get_property):
* gst/interleave/deinterleave.h:
Add a property to select whether channel positions should be kept on
the mono output buffers or should be dropped.
2008-05-19 07:46:05 +00:00
Jan Schmidt
7cf902c4b4 docs/Makefile.am: Oops - fix the spelling of the variable I added.
Original commit message from CVS:
* docs/Makefile.am:
Oops - fix the spelling of the variable I added.
2008-05-18 10:27:25 +00:00
Sebastian Dröge
6827494919 gst/interleave/deinterleave.*: Queue events until src pads were added and they can be sent. Otherwise downstream will...
Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_finalize),
(gst_deinterleave_init), (gst_deinterleave_sink_event),
(gst_deinterleave_process), (gst_deinterleave_sink_activate_push):
* gst/interleave/deinterleave.h:
Queue events until src pads were added and they can be sent. Otherwise
downstream will never get the first newsegment event.
2008-05-17 19:39:53 +00:00
Sebastian Dröge
838e10c219 gst/interleave/deinterleave.c: Always set the channel positions when gst_audio_get_channel_positions() returns someth...
Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_sink_setcaps),
(gst_deinterleave_getcaps):
Always set the channel positions when gst_audio_get_channel_positions()
returns something, even if they're not set in the caps. This makes
sure that the output channels can be interleaved again correctly
in the mono/stereo cases too.
Don't ask for the peercaps of the current pad in getcaps() as this
might call getcaps() again and deadlock.
2008-05-17 14:05:03 +00:00
Sebastian Dröge
67515f39fc ext/timidity/gstwildmidi.c: Check some more common locations for a valid configuration file.
Original commit message from CVS:
* ext/timidity/gstwildmidi.c: (wildmidi_open_config):
Check some more common locations for a valid configuration file.
Fixes bug #533435. Packagers should still #define WILDMIDI_CFG
to the distributions default location.
2008-05-16 22:00:49 +00:00
Sebastian Dröge
5a00cae46f gst/interleave/: Add support for all raw audio formats and provide better negotiation if the caps are changing.
Original commit message from CVS:
* gst/interleave/Makefile.am:
* gst/interleave/deinterleave.c: (deinterleave_24),
(gst_deinterleave_finalize), (gst_deinterleave_base_init),
(gst_deinterleave_class_init), (gst_deinterleave_init),
(gst_deinterleave_add_new_pads), (gst_deinterleave_set_pads_caps),
(gst_deinterleave_set_process_function),
(gst_deinterleave_sink_setcaps), (__remove_channels),
(__set_channels), (gst_deinterleave_getcaps),
(gst_deinterleave_process), (gst_deinterleave_chain),
(gst_deinterleave_sink_activate_push):
* gst/interleave/deinterleave.h:
Add support for all raw audio formats and provide better negotiation
if the caps are changing.
Don't allow changes of the channel positions and set the position of
the corresponding channel on the src pad caps.
General cleanup and smaller bugfixes.
* tests/check/elements/deinterleave.c: (float_buffer_check_probe):
Check the channel positions on the output buffer caps.
2008-05-16 21:56:24 +00:00
Jan Schmidt
d5641e3e6b 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
2008-05-16 19:56:30 +00:00
Jan Schmidt
363d6a2652 docs/plugins/gst-plugins-bad-plugins.types: Remove bogus attempt to pull 'metadata' plugin's base class into the docs.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins.types:
Remove bogus attempt to pull 'metadata' plugin's base
class into the docs.
2008-05-16 14:49:07 +00:00
Wim Taymans
ce2c8b6706 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.
2008-05-14 21:02:19 +00:00
Wim Taymans
affa292951 gst/mpeg4videoparse/mpeg4videoparse.c: Set fixed caps on the srcpad after we created the pad...
Original commit message from CVS:
* gst/mpeg4videoparse/mpeg4videoparse.c: (gst_mpeg4vparse_init):
Set fixed caps on the srcpad after we created the pad...
2008-05-14 20:57:31 +00:00
Tim-Philipp Müller
bfd7046693 tests/check/Makefile.am: Remove deinterleave test from VALGRIND_TO_FIX again now that there are suppressions in gst.s...
Original commit message from CVS:
* tests/check/Makefile.am:
Remove deinterleave test from VALGRIND_TO_FIX again now that
there are suppressions in gst.supp which make this work for me.
2008-05-14 16:21:05 +00:00
Tim-Philipp Müller
426c88452b tests/check/Makefile.am: Add deinterleave unit test to VALGRIND_TO_FIX, since it causes weird invalid free errors in ...
Original commit message from CVS:
* tests/check/Makefile.am:
Add deinterleave unit test to VALGRIND_TO_FIX, since it causes
weird invalid free errors in valgrind/libc after _exit for some
reason.
* tests/check/elements/deinterleave.c: (pads_created),
(set_channel_positions), (src_handoff_float32_8ch),
(float_buffer_check_probe),
(pad_added_setup_data_check_float32_8ch_cb),
(make_fake_src_8chans_float32), (GST_START_TEST),
(deinterleave_suite):
Add some more deinterleave unit test bits I had locally.
2008-05-14 14:19:47 +00:00
Tim-Philipp Müller
8ed88a34d0 gst/audioresample/gstaudioresample.c: Revert previous change which made basetransform handle buffer_alloc and which b...
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
Revert previous change which made basetransform handle buffer_alloc
and which breaks things badly in the non-passthrough case since it
returned buffers with a different (ie. sometimes smaller) size than
the size requested.
2008-05-14 13:57:41 +00:00
Sebastian Dröge
8bcf8c0b0c gst/interleave/: Split definitions into separate header files for better documentation generation.
Original commit message from CVS:
* gst/interleave/Makefile.am:
* gst/interleave/deinterleave.h:
* gst/interleave/interleave.h:
* gst/interleave/plugin.h:
Split definitions into separate header files for better documentation
generation.
* gst/interleave/deinterleave.c: (gst_deinterleave_base_init),
(gst_deinterleave_class_init), (gst_deinterleave_sink_setcaps),
(gst_deinterleave_process):
Don't use alloca, allow caps changes as long as the number of channels
does not change, don't use g_warning, return NOT_NEGOTIATED as early
as possible and some other cleanup.
* gst/interleave/interleave.c: (gst_interleave_base_init),
(gst_interleave_class_init):
Do some random cleanup.
* tests/check/Makefile.am:
* tests/check/elements/deinterleave.c: (GST_START_TEST),
(deinterleave_chain_func), (deinterleave_pad_added),
(deinterleave_suite):
Add unit tests for the deinterleave element.
2008-05-14 07:32:44 +00:00
Sjoerd Simons
7bd28a6a94 gst/mpeg4videoparse/mpeg4videoparse.*: Parse the config data (either outbound or in the stream) to set width/height, ...
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/mpeg4videoparse/mpeg4videoparse.c:
(gst_mpeg4vparse_set_new_caps), (gst_mpeg4vparse_align),
(get_bits), (next_start_code), (gst_mpeg4vparse_handle_vos),
(gst_mpeg4vparse_push), (gst_mpeg4vparse_drain),
(gst_mpeg4vparse_chain), (gst_mpeg4vparse_sink_setcaps),
(gst_mpeg4vparse_sink_event), (gst_mpeg4vparse_src_query),
(gst_mpeg4vparse_set_property), (gst_mpeg4vparse_get_property),
(gst_mpeg4vparse_class_init), (gst_mpeg4vparse_init):
* gst/mpeg4videoparse/mpeg4videoparse.h:
Parse the config data (either outbound or in the stream) to set
width/height, apect ration, framerate in the caps if applicable.
Mark frames as GST_BUFFER_FLAG_DELTA_UNIT when they are not
intra frames
Set the timestamps of outgoing buffers to the buffer in
which the VOP header was found.
Drop incoming data untill configuration is found (by default,
configurable using a property).
Report a 1 frame latency. Fixes #532723.
2008-05-13 17:21:07 +00:00
Wim Taymans
03271f091a gst/real/gstrealvideodec.c: Add some debug for where we are searching for libraries.
Original commit message from CVS:
* gst/real/gstrealvideodec.c: (open_library):
Add some debug for where we are searching for libraries.
2008-05-13 16:16:35 +00:00
Sjoerd Simons
bb04b24eb2 tests/check/elements/audioresample.c: Add unit test for the latest basetransform negotiation changes.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* tests/check/elements/audioresample.c:
(live_switch_alloc_only_48000), (live_switch_get_sink_caps),
(live_switch_push), (GST_START_TEST):
Add unit test for the latest basetransform negotiation changes.
See bug #526768.
2008-05-13 10:59:49 +00:00
Wim Taymans
9fd3fa1063 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.
2008-05-13 09:06:51 +00:00
Wim Taymans
4ab14e0849 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.
2008-05-12 18:43:41 +00:00
Peter Kjellerstedt
11ebf3a6aa 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.
2008-05-12 14:28:09 +00:00
Jan Schmidt
f315cbd74e docs/plugins/gst-plugins-bad-plugins-sections.txt: Quieten some docs output
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
Quieten some docs output
2008-05-12 14:17:06 +00:00
Jan Schmidt
cf965d711c 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.
2008-05-12 14:12:08 +00:00
Jan Schmidt
065dc43a59 Random doc of the day: the deinterlace element.
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:
* docs/plugins/inspect/plugin-gstinterlace.xml:
* gst/deinterlace/gstdeinterlace.c:
* gst/deinterlace/gstdeinterlace.h:
Random doc of the day: the deinterlace element.
2008-05-11 17:23:20 +00:00
Zaheer Abbas Merali
a6e1c40344 gst/mpegtsparse/: Make sure all schedule EIT and non-actual transport stream
Original commit message from CVS:
* gst/mpegtsparse/mpegtspacketizer.c:
* gst/mpegtsparse/mpegtsparse.c:
Make sure all schedule EIT and non-actual transport stream
EITs are parsed. Also add present-following flag and
actual-transport-stream flag to eit bus message.
2008-05-09 10:21:07 +00:00
Peter Kjellerstedt
977c7f0d90 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.
2008-05-09 07:41:58 +00:00
Clive Wright
5d89e2e0a4 sys/oss4/oss4-mixer-slider.c: Apparently mono sliders have the mono value repeated in the upper bits, so mask those o...
Original commit message from CVS:
Based on patch by: Clive Wright <clive_wright ntlworld com>
* sys/oss4/oss4-mixer-slider.c: (gst_oss4_mixer_slider_unpack_volume):
Apparently mono sliders have the mono value repeated in the upper bits,
so mask those out when reading them. Probably makes the mixer applet
work properly in some more cases.
2008-05-08 19:16:17 +00:00
Olivier Crete
f32cbe5017 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.
2008-05-08 09:43:33 +00:00
Sjoerd Simons
08b3343e6b 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.
2008-05-08 06:23:39 +00:00
Sjoerd Simons
61aa25b049 gst/audioresample/gstaudioresample.c: Let audioresample use the buffer allocation of basetransform instead of it's ow...
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
Let audioresample use the buffer allocation of basetransform instead
of it's own stuff.
* tests/check/elements/audioresample.c: (alloc_only_48000),
(GST_START_TEST), (audioresample_suite):
Add unit test for the recent basetransform bugfix, where upstream
changes caps to something that can't be passed through anymore.
2008-05-08 06:20:42 +00:00
Ole André Vadla Ravnås
cfe9a2f6bb win32/common/config.h.in: Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather use the real thing than h...
Original commit message from CVS:
* win32/common/config.h.in:
Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather
use the real thing than having "???" unconditionally.
2008-05-07 20:25:09 +00:00
Wim Taymans
b2aee5ef76 gst-libs/gst/app/: Add marshal.list, make it compile and add to cvsignore.
Original commit message from CVS:
* gst-libs/gst/app/.cvsignore:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/app/gstapp-marshal.list:
Add marshal.list, make it compile and add to cvsignore.
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose),
(gst_app_sink_stop):
Small cleanups.
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
(gst_app_src_init), (gst_app_src_set_property),
(gst_app_src_get_property), (gst_app_src_unlock),
(gst_app_src_unlock_stop), (gst_app_src_start), (gst_app_src_stop),
(gst_app_src_create), (gst_app_src_set_caps),
(gst_app_src_get_caps), (gst_app_src_set_size),
(gst_app_src_get_size), (gst_app_src_set_seekable),
(gst_app_src_get_seekable), (gst_app_src_set_max_buffers),
(gst_app_src_get_max_buffers), (gst_app_src_push_buffer),
(gst_app_src_end_of_stream):
* gst-libs/gst/app/gstappsrc.h:
Beat appsrc in shape, add signals and actions.
Add some docs.
Add properties for caps, size, seekability and max-buffers.
Fix unlock/stop code.
2008-05-07 10:38:23 +00:00
Tim-Philipp Müller
42d7c3409f configure.ac: Error out if we don't have the required versions of core/base.
Original commit message from CVS:
* configure.ac:
Error out if we don't have the required versions of core/base.
2008-05-07 07:51:36 +00:00
Wim Taymans
75695e03a0 gst-libs/gst/app/gstappsink.*: Start some docs.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
(gst_app_sink_init), (gst_app_sink_set_property),
(gst_app_sink_get_property), (gst_app_sink_unlock_start),
(gst_app_sink_unlock_stop), (gst_app_sink_flush_unlocked),
(gst_app_sink_start), (gst_app_sink_stop), (gst_app_sink_event),
(gst_app_sink_preroll), (gst_app_sink_render),
(gst_app_sink_set_caps), (gst_app_sink_set_drop),
(gst_app_sink_get_drop):
* gst-libs/gst/app/gstappsink.h:
Start some docs.
Add property to drop buffers when the queue is filled
Fix unlocking and flushing when the queues are filled.
2008-05-05 10:27:45 +00:00
Christian Schaller
3a959f34ee add wildmidi plugin
Original commit message from CVS:
add wildmidi plugin
2008-05-02 14:40:08 +00:00
Jens Granseuer
9c984b9594 gst/subenc/gstsrtenc.c: Declare variables at the beginning of blocks. Fixes compilation with gcc 2.x and other compil...
Original commit message from CVS:
Patch by: Jens Granseuer <jensgr at gmx dot net>
* gst/subenc/gstsrtenc.c: (gst_srt_enc_timestamp_to_string):
Declare variables at the beginning of blocks. Fixes compilation with
gcc 2.x and other compilers. Fixes bug #530611.
2008-04-29 19:11:56 +00:00
Zaheer Abbas Merali
63fae1c4bb gst/mpegtsparse/: Detect SI pids (NIT, SDT, EIT etc.) based on table id and not by pid number. This allows for exampl...
Original commit message from CVS:
* gst/mpegtsparse/mpegtspacketizer.c:
* gst/mpegtsparse/mpegtspacketizer.h:
* gst/mpegtsparse/mpegtsparse.c:
Detect SI pids (NIT, SDT, EIT etc.) based on table id and not
by pid number. This allows for example the EPG data from UK's
freesat to be picked up.
2008-04-29 09:02:35 +00:00
Sebastian Dröge
625a222f06 ext/: Cast NULL sentinels to void * as NULL is defined as an integer constant in most environments when using C++ and...
Original commit message from CVS:
* ext/mpeg2enc/gstmpeg2enc.cc:
* ext/soundtouch/gstbpmdetect.cc:
Cast NULL sentinels to void * as NULL is defined as an integer
constant in most environments when using C++ and it's size might
be different from a pointer.
2008-04-25 23:22:56 +00:00
Wim Taymans
e4c57a859a gst-libs/gst/app/gstappsink.*: Add more docs.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
(gst_app_sink_init), (gst_app_sink_set_property),
(gst_app_sink_get_property), (gst_app_sink_event),
(gst_app_sink_preroll), (gst_app_sink_render),
(gst_app_sink_set_emit_signals), (gst_app_sink_get_emit_signals),
(gst_app_sink_set_max_buffers), (gst_app_sink_get_max_buffers),
(gst_app_sink_pull_buffer):
* gst-libs/gst/app/gstappsink.h:
Add more docs.
Add signals for when preroll and render buffers are available.
Add property to control signal emission.
Add property to control the max queue size.
2008-04-25 18:18:47 +00:00
Michael Smith
a740294f2a gst-libs/gst/dshow/Makefile.am: Use CXXFLAGS rather than CFLAGS; these are C++ files.
Original commit message from CVS:
* gst-libs/gst/dshow/Makefile.am:
Use CXXFLAGS rather than CFLAGS; these are C++ files.
Define required constants appropriately.
* sys/dshowdecwrapper/Makefile.am:
Add required include dir, libraries.
Define required constants appropriately.
2008-04-25 18:13:07 +00:00
Wim Taymans
1e9fd08525 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.
2008-04-25 11:32:09 +00:00
Wim Taymans
cc6371727c 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.
2008-04-25 11:22:13 +00:00
Wim Taymans
120c55eb27 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.
2008-04-25 09:35:43 +00:00
Wim Taymans
5fe12accc9 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.
2008-04-25 08:21:06 +00:00
Wim Taymans
9285e110cc 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.
2008-04-25 08:15:58 +00:00
Wim Taymans
af19b91216 gst/flv/gstflvdemux.c: Forward unknown queries upstream instead of returning FALSE on them.
Original commit message from CVS:
* gst/flv/gstflvdemux.c: (gst_flv_demux_query):
Forward unknown queries upstream instead of returning FALSE on them.
2008-04-25 08:07:36 +00:00
Sebastian Dröge
01bcb096c7 Add support for the new libmpcdec API which magically gets us support for SV8 files. Also do some random cleanup. Fix...
Original commit message from CVS:
* configure.ac:
* ext/musepack/gstmusepackdec.c: (gst_musepackdec_base_init),
(gst_musepackdec_init), (gst_musepackdec_dispose),
(gst_musepackdec_handle_seek_event), (gst_musepack_stream_init),
(gst_musepackdec_loop), (plugin_init):
* ext/musepack/gstmusepackdec.h:
* ext/musepack/gstmusepackreader.c:
* ext/musepack/gstmusepackreader.h:
Add support for the new libmpcdec API which magically gets us support
for SV8 files. Also do some random cleanup. Fixes bug #526905.
2008-04-24 22:19:48 +00:00
Jan Schmidt
36d15d7868 configure.ac: Back to development -> 0.10.7.1
Original commit message from CVS:
* configure.ac:
Back to development -> 0.10.7.1
=== release 0.10.7 ===
2008-04-24 21:24:18 +00:00
Jan Schmidt
10e4194d5d Release 0.10.7
Original commit message from CVS:
Release 0.10.7
2008-04-24 00:18:30 +00:00
Jan Schmidt
5d68a26ffa Update .po files
Original commit message from CVS:
Update .po files
2008-04-24 00:15:27 +00:00