Original commit message from CVS:
* ChangeLog:
Add missing newline.
* gst/librfb/rfbdecoder.c:
Fix the build (missing stdlib.h).
* gst/spectrum/gstspectrum.c:
* gst/spectrum/gstspectrum.h:
Use basetransform segment so that it is correctly managed on flushes
and start/stop. Report message timestamp as stream time, which is what
an application can understand. (Yes these are adapted from wim recent
level element changes)
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams),
(gst_rtspsrc_dup_printf):
Give meaningfull error when all streams failed to configure for some
reason.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_loop),
(gst_wavparse_chain):
Don't push EOS from the chain function, the element
driving the pipeline is responsible for this. The bug
this was meant to fix seems to be queue not forwarding
EOS in all cases (see #476514).
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_class_init), (gst_level_start),
(gst_level_transform_ip):
* gst/level/gstlevel.h:
Use basetransform segment so that it is correctly managed on flushes and
start/stop.
Report message timestamp as stream time, which is what an application
can understand.
Original commit message from CVS:
* ext/taglib/gstapev2mux.cc:
* ext/taglib/gstapev2mux.h:
* ext/taglib/gsttaglibmux.c:
* tests/check/elements/apev2mux.c:
Update my mail address.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_eos),
(gst_wavparse_loop), (gst_wavparse_chain):
Add EOS logic for the push-based mode too. Fixes#476514.
Original commit message from CVS:
* ext/gconf/gstgconfaudiosink.c:
Fix warning when building without debug.
* sys/oss/gstossmixertrack.c:
Use const like in alsamixertrack.c (fixes warnings).
Original commit message from CVS:
* sys/v4l2/v4l2src_calls.c:
(gst_v4l2src_probe_caps_for_format_and_size):
Fix framerate detection code some more.
Handle the case where there is a weird step in the stepwise framerates.
Don't overwrite the min interval with the framerate, use a temp variable
instead.
Use max in the Continuous framerate intervals instead of step, which is
1 according to the docs. Fixes#475424.
Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_init), (gst_udpsrc_create):
Make udpsrc timestamp outgoing buffers based on when they were received.
Also make it output a segment in time.
Original commit message from CVS:
* configure.ac:
Use AC_TRY_COMPILE instead of AC_TRY_RUN to check for old
flac versions, 's good for cross-compilation karma.
Original commit message from CVS:
Patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com>
* gst/rtp/gstrtph263pay.c:
Fix up header structure so that compilers don't add padding
between the structure fields, since that would lead to us
sending RTP packets with broken headers (as is currently the
case when compiling with MSVC). Also see similar fixes in
libgstrtp in gst-plugins-base. (#474616; #471194)
Original commit message from CVS:
* sys/v4l2/v4l2src_calls.c:
(gst_v4l2src_probe_caps_for_format_and_size):
Don't overwrite our GValue with 0 but instead use the previously
computed value. Fixes#471823 some more.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_start),
(gst_spectrum_transform_ip):
Use the correct parameter order for the memset calls.
Thanks to Christian Schaller for noticing.
Original commit message from CVS:
* docs/plugins/gst-plugins-good-plugins.hierarchy:
No tabs in this file please, or gtk-doc will end up documenting
rather absurd class hierarchies.
Original commit message from CVS:
* ext/gconf/gstswitchsink.c:
If the new kid element fails to change state for some reason
(e.g. esdsink not being able to connect to the sound server),
forward the error message it posted on the bus instead of just
posting a generic 'Internal state change error: please file a
bug' error message. Fixes#471364.
Original commit message from CVS:
* configure.ac:
* gst/spectrum/Makefile.am:
* gst/spectrum/demo-audiotest.c: (draw_spectrum),
(message_handler), (main):
* gst/spectrum/demo-osssrc.c: (draw_spectrum), (message_handler):
* gst/spectrum/gstspectrum.c: (gst_spectrum_base_init),
(gst_spectrum_class_init), (gst_spectrum_init),
(gst_spectrum_dispose), (gst_spectrum_set_property),
(gst_spectrum_get_property), (gst_spectrum_start),
(gst_spectrum_setup), (gst_spectrum_message_new),
(gst_spectrum_transform_ip):
* gst/spectrum/gstspectrum.h:
Port GstSpectrum to GstAudioFilter and libgstfft, add support
for int32, float and double, use floats for the message contents,
average all FFTs done in one interval for better results, use
a better windowing function, allow posting the phase in the message
and actually do an FFT with the requested number of bands instead
of interpolating.
* tests/check/elements/spectrum.c: (GST_START_TEST),
(spectrum_suite):
Improve the units tests by checking for a 11025Hz sine wave
and add unit tests for all 4 supported sample types.
Original commit message from CVS:
* sys/v4l2/gstv4l2src.c:
* sys/v4l2/gstv4l2src.h:
* sys/v4l2/v4l2src_calls.c:
Implement LATENCY queries in the crudest way possible so I don't
have to use sync=false any longer when testing with videosinks.
Original commit message from CVS:
* sys/v4l2/v4l2src_calls.c:
(gst_v4l2src_probe_caps_for_format_and_size):
Add some more debugging in the framerate function.
Iterate stepwise framerate up to and _including_ the max and if nothing
was added to the list, add a dummy 0/1 to 100/1 framerate so that we
don't end up with an empty list.
Original commit message from CVS:
* sys/v4l2/gstv4l2src.c: (gst_v4l2_get_caps_info),
(gst_v4l2src_set_caps), (gst_v4l2src_get_mmap):
Restructure the setcaps function so that we can also compute the
expected GStreamer output size of the video frames.
Set frame_byte_size correctly so that read-based devices have a chance
of working correctly.
When grabbing a frame, discard frames that are not of the expected size.
Some cameras don't output the right framesize for the first buffer.
Try only a couple of times to get a valid frame, else error out.
* sys/v4l2/v4l2_calls.c: (gst_v4l2_get_capabilities),
(gst_v4l2_fill_lists), (gst_v4l2_get_input):
Add some more debug info when scanning the device.
* sys/v4l2/v4l2src_calls.c: (gst_v4l2_buffer_new),
(gst_v4l2_buffer_pool_new), (gst_v4l2_buffer_pool_activate),
(gst_v4l2src_fill_format_list), (gst_v4l2src_grab_frame),
(gst_v4l2src_set_capture), (gst_v4l2src_capture_init):
Add some more debug info when dequeing a frame.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c:
* gst/wavparse/gstwavparse.h:
Implement seek-query. Refactor duration calculations. Appropriate use
of uint64_scale_int and uint64_scale. Move repeadedly calculated stuff
out of loops.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_udp_sink),
(gst_rtspsrc_dup_printf):
Use new basesink async property to make sparse RTCP packet not wait for
preroll.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_dup_printf),
(gst_rtspsrc_get_float), (gst_rtspsrc_play):
Make sure we generate and parse floating point values in the POSIX
locale instead of the current locale.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_seek),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
(gst_rtspsrc_play):
* gst/rtsp/gstrtspsrc.h:
Fix method detection again.
Keep track of when we must send a Range header.
Use segment values for Range, Speed and Scale headers.
Parse Speed and Scale headers to update the segment values.
Original commit message from CVS:
* gst/audiofx/audiochebyshevfreqband.c:
* gst/audiofx/audiochebyshevfreqlimit.c:
Add small comparision with the windowed sinc filters in the docs.
Original commit message from CVS:
* tests/check/elements/bpwsinc.c: (GST_START_TEST),
(bpwsinc_suite):
* tests/check/elements/lpwsinc.c: (GST_START_TEST),
(lpwsinc_suite):
Also test everything in 32 bit float mode.
Original commit message from CVS:
* tests/check/elements/audiochebyshevfreqband.c: (GST_START_TEST),
(audiochebyshevfreqband_suite):
* tests/check/elements/audiochebyshevfreqlimit.c: (GST_START_TEST),
(audiochebyshevfreqlimit_suite):
Also test 32 bit float mode and the type 2 variants of the filters.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
(gst_rtspsrc_loop):
Refactor the udp and interleaved loop function a bit.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_connection_send),
(gst_rtspsrc_connection_receive), (gst_rtspsrc_sink_chain),
(gst_rtspsrc_handle_request), (gst_rtspsrc_send_keep_alive),
(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
(gst_rtspsrc_try_send), (gst_rtspsrc_pause):
* gst/rtsp/gstrtspsrc.h:
Protect connection activity with a new lock, avoids deadlocks when going
to PAUSED. Fixes#455808.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
(bpwsinc_set_property), (bpwsinc_get_property):
* gst/filter/gstbpwsinc.h:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
(gst_lpwsinc_init), (lpwsinc_build_kernel), (lpwsinc_set_property),
(lpwsinc_get_property):
* gst/filter/gstlpwsinc.h:
* tests/check/elements/lpwsinc.c: (GST_START_TEST):
Use generator macros for the process functions for the different
sample types, add lower upper boundaries for the GObject properties
so automatically generated UIs can use sliders and change frequency
properties to floats to save a bit of memory, even ints would in
theory be enough. Also rename frequency to cutoff for consistency
reasons.
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
Regenerated for the above changes.
Original commit message from CVS:
* gst/audiofx/audiochebyshevfreqband.c:
(gst_audio_chebyshev_freq_band_class_init):
* gst/audiofx/audiochebyshevfreqlimit.c:
(gst_audio_chebyshev_freq_limit_class_init):
Use generator macros for the process functions for the different
sample types, add lower upper boundaries for the GObject properties
so automatically generated UIs can use sliders and add a note about
the number of poles as a too high number of poles combined with
very low or very high frequencies will produce only noise.
* docs/plugins/gst-plugins-good-plugins.args:
Regenerated for the property changes.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_property),
(gst_rtspsrc_flush), (gst_rtspsrc_sink_chain),
(gst_rtspsrc_stream_configure_udp_sink),
(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
(gst_rtspsrc_try_send), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_parse_range),
(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_pause),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Improve timeout handling.
Use the same socket for sending and receiving RTCP packets so that some
servers can track clients better.
Improve connection closed handling. Try to reconnect.
Don't overwrite our content base with NULL.
Improve debugging.
Improve range parsing and handling.
Remove flushing hack now that core does the right thing.
Original commit message from CVS:
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
(gst_multiudpsink_init), (gst_multiudpsink_set_property),
(gst_multiudpsink_get_property), (gst_multiudpsink_init_send),
(gst_multiudpsink_close), (gst_multiudpsink_add):
* gst/udp/gstmultiudpsink.h:
Add support for getting and setting the socket to use.
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
(gst_udpsrc_create), (gst_udpsrc_get_property):
Add support for getting the currently used socket.
