Original commit message from CVS:
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
(gst_rtp_h264_depay_init), (gst_rtp_h264_depay_set_property),
(gst_rtp_h264_depay_get_property), (gst_rtp_h264_depay_setcaps),
(gst_rtp_h264_depay_process):
* gst/rtp/gstrtph264depay.h:
Add experimental support for outputting quicktime-like AVC output in
addition to the existing bytestream output.
* gst/rtp/gstrtph264pay.c: (gst_h264_scan_mode_get_type),
(gst_rtp_h264_pay_class_init), (gst_rtp_h264_pay_init),
(gst_rtp_h264_pay_setcaps), (gst_rtp_h264_pay_payload_nal),
(gst_rtp_h264_pay_handle_buffer), (gst_rtp_h264_pay_set_property),
(gst_rtp_h264_pay_get_property):
* gst/rtp/gstrtph264pay.h:
Make the parsing mode configurable, for some inputs we don't need to
scan every byte for start codes.
Only set the marker bit on ACCESS units.
Original commit message from CVS:
* gst/equalizer/gstiirequalizer.c:
Use a bigger type in integer mode for the intermediate results to
prevent overflows. This fixes the crippled sound when using the
equalizer in integer mode. Fixes bug #510865.
Original commit message from CVS:
* gst/videomixer/videomixer.c:
* gst/videomixer/videomixer.h:
Instead of a random number for the request pad id's,
use a counter.
Register the videomixerpad class from the element's class_init
where it's safer, and allows the docs generator to scan it.
Original commit message from CVS:
* gst/smpte/Makefile.am:
* gst/smpte/gstsmpte.c: (gst_smpte_plugin_init):
* gst/smpte/gstsmpte.h:
* gst/smpte/gstsmptealpha.c:
(gst_smpte_alpha_transition_type_get_type),
(gst_smpte_alpha_get_type), (gst_smpte_alpha_base_init),
(gst_smpte_alpha_class_init), (gst_smpte_alpha_update_mask),
(gst_smpte_alpha_setcaps), (gst_smpte_alpha_get_unit_size),
(gst_smpte_alpha_init), (gst_smpte_alpha_finalize),
(gst_smpte_alpha_do_ayuv), (gst_smpte_alpha_do_i420),
(gst_smpte_alpha_transform), (gst_smpte_alpha_set_property),
(gst_smpte_alpha_get_property), (gst_smpte_alpha_plugin_init):
* gst/smpte/gstsmptealpha.h:
* gst/smpte/plugin.c: (plugin_init):
Add new plugin that adds the SMPTE transition in the alpha channel of
I420 and AYUV frames so that they can be blended with videomixer later
on. Uses all niceties such as using base transform for efficient alloc
and negotiation. It currently requires GstController to control the
position in the transition effect.
Original commit message from CVS:
* 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/gst-plugins-good-plugins.hierarchy:
* docs/plugins/gst-plugins-good-plugins.interfaces:
* docs/plugins/gst-plugins-good-plugins.types:
* gst/videomixer/videomixer.c:
Try using thaytans new mechanism to get extra classes into plugin
docs. Aparently works for the Eq. For VideoMixer the GObject stuff is
missing still.
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.
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.
Original commit message from CVS:
* gst/avi/gstavimux.c: (gst_avi_mux_audsink_set_caps):
Set proper rate in avi stream header for PCM audio, and also do some
more sanity checks on caps in this case. Fixes#511489.
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.
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.
Original commit message from CVS:
* sys/v4l2/gstv4l2src.c:
Don't include the gstv4l2xoverlay.h header as the XOverlay support
isn't implemented at all yet and this requires X headers to be
installed. Fixes bug #533264.
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.
Original commit message from CVS:
* ext/wavpack/gstwavpackstreamreader.c:
* tests/examples/spectrum/demo-audiotest.c:
* tests/examples/spectrum/demo-osssrc.c:
Fix some compiler warnings.
Original commit message from CVS:
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_process):
Small comment added.
* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_class_init),
(gst_rtp_h264_pay_decode_nal), (gst_rtp_h264_pay_parse_sps_pps),
(gst_rtp_h264_pay_payload_nal), (gst_rtp_h264_pay_handle_buffer):
Debug string cleanups (remove trailing \n)
Refactor and clean up the payloader a bit and make sure that we only
put one NAL unit in an RTP packet even if the input buffer contains
multiple NAL units.
Add suport for AVC format input.
Original commit message from CVS:
* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_class_init),
(gst_rtp_h264_pay_finalize), (gst_rtp_h264_pay_handle_buffer),
(gst_rtp_h264_pay_set_property), (gst_rtp_h264_pay_get_property):
* gst/rtp/gstrtph264pay.h:
Make it possible to specify profile-level-id and sprop-parameter-sets
using properties in case they are not available in-stream.
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.
Original commit message from CVS:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/inspect/plugin-ladspa.xml:
Remove ladspa fro plugin-docs, its in gst-plugins-bad.
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.
Original commit message from CVS:
* gst/avi/gstavimux.c: (gst_avi_mux_start_file):
Send an initial BYTE segment to inform downstream of later seeking,
and to forego sync attempts.
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_push_dvd_clut_change_event):
Convert subtitle palette info in VobSub private data from VobSub's
(buggy) RGB to YUV.
Original commit message from CVS:
* gst/avi/gstavimux.c: (gst_avi_mux_pad_reset):
Do not leave fourcc stream header field empty upon reset.
Fixes#519301.
Original commit message from CVS:
Based on patch by: Wouter Cloetens <wouter at mind be>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_skip_lws), (gst_rtspsrc_unskip_lws),
(gst_rtspsrc_skip_commas), (gst_rtspsrc_skip_item),
(gst_rtsp_decode_quoted_string),
(gst_rtspsrc_parse_digest_challenge), (gst_rtspsrc_parse_auth_hdr),
(gst_rtspsrc_setup_auth):
Support Digest authentication. Fixes#532065.
Original commit message from CVS:
* gst/level/gstlevel.c:
Also support 32bit (e.g. whe having it after 'mad'). Add more notes
about whats needed for liboil acceleration. Simplify docs a bit.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/matroska/matroska-mux.c: (gst_matroska_mux_collected):
Update the track duration if the old one was invalid.
Fixes bug #532117.
Original commit message from CVS:
* gst/rtp/gstrtph264pay.c (gst_rtp_h264_pay_parse_sps_pps):
Use GST_STR_NULL when trying to print sps and pps strings that could
be NULL, as this might crash on some platforms.
Original commit message from CVS:
patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com>
* sys/directdraw/gstdirectdrawsink.c (gst_directdraw_sink_setup_ddraw):
Do IDirectDrawClipper_SetHWnd() if the window ID has already been
set after creating the clipper.
Original commit message from CVS:
patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com>
* sys/directdraw/gstdirectdrawsink.c (gst_directdraw_sink_show_frame):
Added checking of surface lost case after an unsuccessful
IDirectDrawSurface7_Lock() call.
If surface is lost, return GST_FLOW_OK.
Original commit message from CVS:
patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com>
* sys/directdraw/gstdirectdrawsink.c (gst_directdraw_sink_show_frame,
WndProc, gst_directdraw_sink_window_thread):
Improved Windows message loop and fixed window destruction issue.
When the window which DirectDraw is rendering to is destroyed, the
render/show_frame function will return GST_FLOW_ERROR.
Partially fixes#520885.
Original commit message from CVS:
patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com>
* sys/directdraw/gstdirectdrawsink.c (gst_directdraw_sink_set_caps):
Fixed mid stream resolution change bug, the offscreen surface is now
released when set_caps is called.
Partially fixes#520885.
Original commit message from CVS:
* sys/directdraw/gstdirectdrawsink.c
(gst_directdraw_sink_buffer_alloc):
Make it so that gst_directdraw_sink_buffer_alloc uses the right
width/height.
Especially when looking through the pool of buffers, make sure that
the width/height of caps is used instead of the already negotiated
dimensions.
For example if a buffer with different caps is requested, i.e.
higher resolution, the caller would get a buffer with the old
dimensions and thus corrupt the heap.
Original commit message from CVS:
* sys/directdraw/gstdirectdrawsink.c
(gst_directdraw_sink_buffer_alloc):
Clear the flags on recycled buffers from buffer_alloc.
Partially fixes#520885.
The right fix this time.
Original commit message from CVS:
* sys/directdraw/gstdirectdrawsink.c
(gst_directdraw_sink_buffer_alloc):
Reverting previous commit, it had it all mixed up, was for a different
patch (major automation screw-up). Sorry!
Original commit message from CVS:
* sys/directdraw/gstdirectdrawsink.c
(gst_directdraw_sink_buffer_alloc):
Clear the flags on recycled buffers from buffer_alloc.
Partially fixes#520885.
Original commit message from CVS:
* gst/rtp/gstrtpilbcpay.c:
Added missing stdlib.h include for strtol(), and made include ordering and
style consistent with the corresponding depayloader.
Original commit message from CVS:
* gst/rtp/gstrtpilbcpay.c:
Added missing stdlib.h include for strtol(), and made include ordering and
style consistent with the corresponding depayloader.
Original commit message from CVS:
* ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_reset),
(gst_gconf_audio_src_change_state):
* ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_reset),
(gst_gconf_video_sink_change_state):
* ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_reset),
(gst_gconf_video_src_change_state):
* ext/gconf/gstswitchsink.c: (gst_switch_sink_reset),
(gst_switch_commit_new_kid), (gst_switch_sink_change_state):
When we can't create a fakesink/fakesrc complain instead of unreffing
NULL pointers and crashing later. See bug #530535.
Original commit message from CVS:
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_process):
Add some more debug info and guard against small payloads.
* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_process):
Set duration on outgoing buffers because we can.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* ext/speex/gstspeexenc.c: (gst_speex_enc_sink_getcaps),
(gst_speex_enc_init), (gst_speex_enc_chain):
Add negotiation for the speex channels and rate. Fixes#465146.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_class_init),
(gst_rtp_speex_pay_getcaps):
Add negotiation for the speec channels and rate. See #465146.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_class_init),
(gst_rtpilbcpay_sink_setcaps), (gst_rtpilbcpay_sink_getcaps):
Add negotiation for the ILBC mode. See #465146.
