ext/gconf/gstswitchsink.c: If the new kid element fails to change state for some reason forward the error message it ...

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.

Port GstSpectrum to GstAudioFilter and libgstfft, add support for int32, float and double, use floats for the message...

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:
* gst/qtdemux/Makefile.am:
* gst/qtdemux/qtdemux.c:
Don't assume tags are encoded as UTF-8 (#473670).

sys/v4l2/: Implement LATENCY queries in the crudest way possible so I don't have to use sync=false any longer when te...

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:
* configure.ac:
Fix build.

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:
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
(gst_multiudpsink_set_clients_string),
(gst_multiudpsink_get_clients_string),
(gst_multiudpsink_set_property), (gst_multiudpsink_get_property),
(gst_multiudpsink_init_send), (gst_multiudpsink_add_internal),
(gst_multiudpsink_add), (gst_multiudpsink_clear_internal),
(gst_multiudpsink_clear):
Add property do configure destination address/port pairs
API:GstMultiUDPSink::clients

Original commit message from CVS:
* tests/examples/Makefile.am:
* tests/examples/rtp/Makefile.am:
* tests/examples/rtp/client-H263p-AMR.sh:
* tests/examples/rtp/client-H263p-PCMA.sdp:
* tests/examples/rtp/client-H263p-PCMA.sh:
* tests/examples/rtp/client-H264-PCMA.sdp:
* tests/examples/rtp/client-H264-PCMA.sh:
* tests/examples/rtp/client-PCMA.sh:
* tests/examples/rtp/server-VTS-H263p-ATS-PCMA.sh:
* tests/examples/rtp/server-alsasrc-PCMA.sh:
* tests/examples/rtp/server-v4l2-H263p-alsasrc-AMR.sh:
* tests/examples/rtp/server-v4l2-H264-alsasrc-PCMA.sh:
Added some RTP example scripts for sending and receiving RTP streams.

sys/v4l2/gstv4l2src.c: Restructure the setcaps function so that we can also compute the expected GStreamer output siz...

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:
More code cleanups. Add some more comment and improve debugs logs.

gst/wavparse/gstwavparse.*: Implement seek-query. Refactor duration calculations. Appropriate use of uint64_scale_int...

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/avi/gstavidemux.c:
Implement seek-query.

gst/rtsp/gstrtspsrc.c: 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_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/audiofx/Makefile.am:
Dist the right file.

gst/rtsp/gstrtspsrc.c: Make sure we generate and parse floating point values in the POSIX locale instead of the curre...

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:
patch by: Mark Nauwelaerts <manauw@skynet.be>
* sys/v4l2/v4l2src_calls.c:
Handle optional v4l2 ioctls gracefully.

Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_base_init),
(gst_rtp_h263_depay_class_init), (gst_rtp_h263_depay_init),
(gst_rtp_h263_depay_finalize), (gst_rtp_h263_depay_setcaps),
(gst_rtp_h263_depay_process), (gst_rtp_h263_depay_set_property),
(gst_rtp_h263_depay_get_property),
(gst_rtp_h263_depay_change_state),
(gst_rtp_h263_depay_plugin_init):
* gst/rtp/gstrtph263depay.h:
Added an H263 depayloader. Fixes #369392.
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps),
(gst_rtp_h263p_depay_process):
* gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type),
(gst_rtp_h263p_pay_class_init), (gst_rtp_h263p_pay_flush):
Make the H263+ pay/depayloader support H263-1998 and H263-2000
payloads.
Also alow plain H263 on the h263p payloaders. Fixes #465040.

Original commit message from CVS:
* gst/filter/gstbpwsinc.c:
* gst/filter/gstlpwsinc.c:
Add small comparision with the chebyshev filters in the docs.

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.

gst/rtsp/gstrtspsrc.*: Protect connection activity with a new lock, avoids deadlocks when going to PAUSED. Fixes #455...

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/debug/rndbuffersize.c: (gst_rnd_buffer_size_loop):
Fix debug statement.

Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_stream_eos):
Fix stray %u in debug line as spotted by Saur on IRC.

Use generator macros for the process functions for the different sample types, add lower upper boundaries for the GOb...

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.

gst/audiofx/: Use generator macros for the process functions for the different sample types, add lower upper boundari...

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.

gst/filter/gstbpwsinc.*: Implement latency query and only forward those samples downstream that actually contain the ...

Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
(gst_bpwsinc_init), (process_32), (process_64),
(bpwsinc_build_kernel), (bpwsinc_push_residue),
(bpwsinc_transform), (bpwsinc_start), (bpwsinc_query),
(bpwsinc_query_type), (bpwsinc_event), (bpwsinc_set_property):
* gst/filter/gstbpwsinc.h:
Implement latency query and only forward those samples downstream
that actually contain the data we want, i.e. drop kernel_length/2
in the beginning and append kernel_length/2 (created by convolving
the filter kernel with zeroes) to the end.
* tests/check/elements/bpwsinc.c: (GST_START_TEST):
Adjust the unit test for this slightly changed behaviour.
* gst/filter/gstlpwsinc.c: (lpwsinc_build_kernel):
Reset residue length only when actually creating a residue.

Original commit message from CVS:
reviewed by: Stefan Kost  <ensonic@users.sf.net>
* gst/audiofx/Makefile.am:
* gst/audiofx/audiochebyshevfreqband.c:
(gst_audio_chebyshev_freq_band_mode_get_type),
(gst_audio_chebyshev_freq_band_base_init),
(gst_audio_chebyshev_freq_band_dispose),
(gst_audio_chebyshev_freq_band_class_init),
(gst_audio_chebyshev_freq_band_init),
(generate_biquad_coefficients), (calculate_gain),
(generate_coefficients),
(gst_audio_chebyshev_freq_band_set_property),
(gst_audio_chebyshev_freq_band_get_property),
(gst_audio_chebyshev_freq_band_setup), (process), (process_64),
(process_32), (gst_audio_chebyshev_freq_band_transform_ip),
(gst_audio_chebyshev_freq_band_start):
* gst/audiofx/audiochebyshevfreqband.h:
* gst/audiofx/audiochebyshevfreqlimit.c:
(gst_audio_chebyshev_freq_limit_mode_get_type),
(gst_audio_chebyshev_freq_limit_base_init),
(gst_audio_chebyshev_freq_limit_dispose),
(gst_audio_chebyshev_freq_limit_class_init),
(gst_audio_chebyshev_freq_limit_init),
(generate_biquad_coefficients), (calculate_gain),
(generate_coefficients),
(gst_audio_chebyshev_freq_limit_set_property),
(gst_audio_chebyshev_freq_limit_get_property),
(gst_audio_chebyshev_freq_limit_setup), (process), (process_64),
(process_32), (gst_audio_chebyshev_freq_limit_transform_ip),
(gst_audio_chebyshev_freq_limit_start):
* gst/audiofx/audiochebyshevfreqlimit.h:
* gst/audiofx/audiofx.c: (plugin_init):
Add Chebyshev lowpass/highpass and bandpass/bandreject elements.
Fixes #464800.
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/audiochebyshevfreqband.c:
(setup_audiochebyshevfreqband), (cleanup_audiochebyshevfreqband),
(GST_START_TEST), (audiochebyshevfreqband_suite), (main):
* tests/check/elements/audiochebyshevfreqlimit.c:
(setup_audiochebyshevfreqlimit), (cleanup_audiochebyshevfreqlimit),
(GST_START_TEST), (audiochebyshevfreqlimit_suite), (main):
Add unit tests for the chebyshev filters.
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/inspect/plugin-1394.xml:
* docs/plugins/inspect/plugin-audiofx.xml:
* docs/plugins/inspect/plugin-dv.xml:
* docs/plugins/inspect/plugin-flac.xml:
* docs/plugins/inspect/plugin-jpeg.xml:
* docs/plugins/inspect/plugin-png.xml:
* docs/plugins/inspect/plugin-rtp.xml:
* docs/plugins/inspect/plugin-shout2send.xml:
* docs/plugins/inspect/plugin-wavpack.xml:
And add docs for the chebyshev filters. While doing
that also run make update in docs/plugins.

Original commit message from CVS:
* ext/annodex/gstcmmltag.c:
* gst/rtp/gstrtpvorbispay.c:
Make ro memory to share.

gst/udp/gstudpsrc.c: Improve UDP performance by avoiding a select() when we have data available immediatly.

Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
Improve UDP performance by avoiding a select() when we have data
available immediatly.

Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_VOID__UINT_UINT),
(gst_rtp_dec_class_init):
* gst/rtsp/gstrtpdec.h:
Add (dummy) SSRC management signals.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
(find_stream), (gst_rtspsrc_create_stream), (new_session_pad),
(request_pt_map), (gst_rtspsrc_do_stream_eos), (on_bye_ssrc),
(on_timeout), (gst_rtspsrc_stream_configure_manager),
(gst_rtspsrc_stream_push_event), (gst_rtspsrc_push_event),
(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_rtpinfo),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Add connection-speed property.
Add find_stream helper functions.
Handle stream EOS based on BYE messages or SSRC timeout.
Returns SUCCESS from the state change function as we hide our async
elements from the parent.

gst/filter/gstlpwsinc.*: Implement latency query and only forward those samples downstream that actually contain the ...

Original commit message from CVS:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
(gst_lpwsinc_init), (process_32), (process_64),
(lpwsinc_build_kernel), (lpwsinc_push_residue),
(lpwsinc_transform), (lpwsinc_start), (lpwsinc_query),
(lpwsinc_query_type), (lpwsinc_event), (lpwsinc_set_property):
* gst/filter/gstlpwsinc.h:
Implement latency query and only forward those samples downstream
that actually contain the data we want, i.e. drop kernel_length/2
in the beginning and append kernel_length/2 (created by convolving
the filter kernel with zeroes) to the end.
* tests/check/elements/lpwsinc.c: (GST_START_TEST):
Adjust the unit test for this slightly changed behaviour.

Original commit message from CVS:
* gst/debug/rndbuffersize.c:
Fix da leak.

Original commit message from CVS:
* gst/debug/Makefile.am:
* gst/debug/breakmydata.c:
* gst/debug/gstdebug.c:
* gst/debug/negotiation.c:
* gst/debug/progressreport.c:
* gst/debug/rndbuffersize.c:
* gst/debug/testplugin.c:
Add new test element and clean-up the others a little.

Add docs for lpwsinc and bpwsinc and integrate them into the build system. While doing that also update all other doc...

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/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-bz2.xml:
* docs/plugins/inspect/plugin-cdxaparse.xml:
* docs/plugins/inspect/plugin-dtsdec.xml:
* docs/plugins/inspect/plugin-faac.xml:
* docs/plugins/inspect/plugin-faad.xml:
* docs/plugins/inspect/plugin-filter.xml:
* docs/plugins/inspect/plugin-freeze.xml:
* docs/plugins/inspect/plugin-gsm.xml:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
* docs/plugins/inspect/plugin-h264parse.xml:
* docs/plugins/inspect/plugin-modplug.xml:
* docs/plugins/inspect/plugin-mpeg2enc.xml:
* docs/plugins/inspect/plugin-musepack.xml:
* docs/plugins/inspect/plugin-musicbrainz.xml:
* docs/plugins/inspect/plugin-nsfdec.xml:
* docs/plugins/inspect/plugin-replaygain.xml:
* docs/plugins/inspect/plugin-soundtouch.xml:
* docs/plugins/inspect/plugin-spcdec.xml:
* docs/plugins/inspect/plugin-spectrum.xml:
* docs/plugins/inspect/plugin-speed.xml:
* docs/plugins/inspect/plugin-tta.xml:
* docs/plugins/inspect/plugin-videosignal.xml:
* docs/plugins/inspect/plugin-xingheader.xml:
* docs/plugins/inspect/plugin-xvid.xml:
* gst/filter/gstbpwsinc.c:
* gst/filter/gstbpwsinc.h:
* gst/filter/gstlpwsinc.c:
* gst/filter/gstlpwsinc.h:
Add docs for lpwsinc and bpwsinc and integrate them
into the build system. While doing that also update
all other docs via make update in docs/plugins.

Original commit message from CVS:
* tests/check/elements/bpwsinc.c: (GST_START_TEST):
Make one test constraint a bit stricter.