Edward Hervey [Thu, 3 May 2007 10:47:22 +0000 (10:47 +0000)]
gst/videorate/gstvideorate.c: There is no sensible way to handle incoming buffers which don't have a valid timestamp....
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_video_rate_chain):
There is no sensible way to handle incoming buffers which don't have a
valid timestamp. We therefore discard them and wait for the next one.
Tim-Philipp Müller [Tue, 1 May 2007 18:45:36 +0000 (18:45 +0000)]
gst/playback/: Better error message for text files.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (type_found), (plugin_init):
* gst/playback/gstdecodebin2.c: (plugin_init):
Better error message for text files.
Wim Taymans [Sun, 29 Apr 2007 14:38:05 +0000 (14:38 +0000)]
gst-libs/gst/rtp/gstrtcpbuffer.c: Fix offset bug in generation RR packets.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_add_rb):
Fix offset bug in generation RR packets.
Julien Moutte [Fri, 27 Apr 2007 15:33:46 +0000 (15:33 +0000)]
ext/theora/theoradec.c: Calculate buffer duration correctly to generate a perfect stream (#433888).
Original commit message from CVS:
2007-04-27 Julien MOUTTE <julien@moutte.net>
* ext/theora/theoradec.c: (_theora_granule_time),
(theora_dec_push_forward), (theora_handle_data_packet),
(theora_dec_decode_buffer): Calculate buffer duration correctly
to generate a perfect stream (#433888).
* gst/audioresample/gstaudioresample.c:
(audioresample_check_discont): Glib provides ABS.
Wim Taymans [Fri, 27 Apr 2007 15:01:40 +0000 (15:01 +0000)]
gst-libs/gst/rtp/gstrtcpbuffer.*: Fix RB block parsing and writing.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
(gst_rtcp_packet_add_rb), (gst_rtcp_packet_sdes_add_item),
(gst_rtcp_packet_sdes_add_entry), (gst_rtcp_packet_bye_add_ssrc),
(gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
(gst_rtcp_packet_bye_set_reason):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Fix RB block parsing and writing.
Add support for constructing BYE packets.
Tim-Philipp Müller [Wed, 25 Apr 2007 08:54:34 +0000 (08:54 +0000)]
When posting a warning message because samples were dropped, post something more intelligible than he default error m...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init),
(gst_base_audio_src_create):
* po/POTFILES.in:
When posting a warning message because samples were dropped, post
something more intelligible than he default error message for clock
errors which is just confusing in this context (#432984).
Wim Taymans [Wed, 25 Apr 2007 08:10:26 +0000 (08:10 +0000)]
gst-libs/gst/rtp/gstrtcpbuffer.*: Implement code to write SR, RR and SDES packets.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_buffer_new),
(gst_rtcp_buffer_end), (gst_rtcp_buffer_get_packet_count),
(read_packet_header), (gst_rtcp_packet_move_to_next),
(gst_rtcp_buffer_add_packet), (gst_rtcp_packet_sr_set_sender_info),
(gst_rtcp_packet_rr_set_ssrc), (gst_rtcp_packet_add_rb),
(gst_rtcp_packet_sdes_get_item_count),
(gst_rtcp_packet_sdes_first_item),
(gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_ssrc),
(gst_rtcp_packet_sdes_first_entry),
(gst_rtcp_packet_sdes_next_entry),
(gst_rtcp_packet_sdes_get_entry), (gst_rtcp_packet_sdes_add_item),
(gst_rtcp_packet_sdes_add_entry):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Implement code to write SR, RR and SDES packets.
Christian Kirbach [Tue, 24 Apr 2007 20:45:24 +0000 (20:45 +0000)]
sys/ximage/ximagesink.c: Fix build if XShm is not available (#432362).
Original commit message from CVS:
Patch by: Christian Kirbach <Christian dot Kirbach at googlemail com>
* sys/ximage/ximagesink.c:
Fix build if XShm is not available (#432362).
Sebastian Dröge [Tue, 24 Apr 2007 18:58:25 +0000 (18:58 +0000)]
gst/audioconvert/gstaudioconvert.c: Initalize the AudioConvertCtx with zeroes, otherwise it will contain pointers to ...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init):
Initalize the AudioConvertCtx with zeroes, otherwise it will contain
pointers to random memory which are passed to g_free() when
audio_convert_prepare_context() is called the first time.
Dan Williams [Tue, 24 Apr 2007 15:00:07 +0000 (15:00 +0000)]
gst/videorate/gstvideorate.c: Don't leak incoming buffer if gst_pad_push() returns a non-OK flow. Fixes #432755.
Original commit message from CVS:
Patch by: Dan Williams <dcbw redhat com>
* gst/videorate/gstvideorate.c: (gst_video_rate_chain):
Don't leak incoming buffer if gst_pad_push() returns a
non-OK flow. Fixes #432755.
* tests/check/elements/videorate.c: (GST_START_TEST),
(videorate_suite):
Unit test for the above by Yours Truly.
Stefan Kost [Mon, 23 Apr 2007 20:04:28 +0000 (20:04 +0000)]
gst/adder/gstadder.c: Fix non-flushing segmented seeks, Fixes #340060 for me
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
(gst_adder_sink_event), (gst_adder_collected):
Fix non-flushing segmented seeks, Fixes #340060 for me
Tim-Philipp Müller [Sat, 21 Apr 2007 15:29:27 +0000 (15:29 +0000)]
ChangeLog surgery: add API keyword
Original commit message from CVS:
ChangeLog surgery: add API keyword
Olivier Crete [Sat, 21 Apr 2007 15:25:22 +0000 (15:25 +0000)]
gst-libs/gst/rtp/gstbasertpaudiopayload.c: Chain up to parent class in dispose function; get rid of unnecessary 'dipo...
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester ca>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_class_init),
(gst_base_rtp_audio_payload_init),
(gst_base_rtp_audio_payload_dispose):
Chain up to parent class in dispose function; get rid of
unnecessary 'diposed' flag in private structure (#415001).
Tim-Philipp Müller [Sat, 21 Apr 2007 15:10:25 +0000 (15:10 +0000)]
Some minor docs fixes and additions; also add missing 'Since' bits.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs.types:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_class_init):
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertppayload.c:
Some minor docs fixes and additions; also add missing 'Since' bits.
Zeeshan Ali [Sat, 21 Apr 2007 14:40:45 +0000 (14:40 +0000)]
gst-libs/gst/rtp/gstbasertpaudiopayload.*: The recently-added gst_base_rtp_audio_payload_push() should take an object...
Original commit message from CVS:
Patch by: Zeeshan Ali <zeenix gmail com>
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_handle_frame_based_buffer),
(gst_base_rtp_audio_payload_handle_sample_based_buffer),
(gst_base_rtp_audio_payload_push):
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
The recently-added gst_base_rtp_audio_payload_push() should take an
object of type GstBaseRTPAudioPayload as first argument (#431672).
Tim-Philipp Müller [Sat, 21 Apr 2007 14:14:24 +0000 (14:14 +0000)]
gst/audioresample/gstaudioresample.c: Make more functions static, just because we can.
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
Make more functions static, just because we can.
Tim-Philipp Müller [Sat, 21 Apr 2007 13:54:39 +0000 (13:54 +0000)]
tests/check/elements/audioresample.c: Add unit test for audioresample shutdown crasher (#420106).
Original commit message from CVS:
* tests/check/elements/audioresample.c:
Add unit test for audioresample shutdown crasher (#420106).
Stefan Kost [Fri, 20 Apr 2007 10:42:24 +0000 (10:42 +0000)]
gst/subparse/: Use GST_DISABLE_XML here
Original commit message from CVS:
* gst/subparse/gstsubparse.c:
* gst/subparse/samiparse.c:
Use GST_DISABLE_XML here
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
(gst_xvimagesink_xwindow_new), (gst_xvimagesink_get_xv_support),
(gst_xvimagesink_buffer_alloc),
(gst_xvimagesink_navigation_send_event):
* sys/xvimage/xvimagesink.h:
Include stdlib.h when using atoi.
* tests/check/elements/playbin.c: (playbin_suite):
Use GST_DISABLE_REGISTRY here
Michael Smith [Thu, 19 Apr 2007 16:58:53 +0000 (16:58 +0000)]
ext/theora/: Track initialisation state; don't try to use encoder state if we're not initialised (it'll segfault).
Original commit message from CVS:
* ext/theora/gsttheoraenc.h:
* ext/theora/theoraenc.c: (theora_enc_sink_setcaps),
(theora_enc_sink_event), (theora_enc_change_state):
Track initialisation state; don't try to use encoder state if we're
not initialised (it'll segfault).
Stefan Kost [Wed, 18 Apr 2007 11:06:42 +0000 (11:06 +0000)]
tests/check/pipelines/.cvsignore: Fix build.
Original commit message from CVS:
* tests/check/pipelines/.cvsignore:
Fix build.
Tim-Philipp Müller [Tue, 17 Apr 2007 10:56:37 +0000 (10:56 +0000)]
gst/app/Makefile.am: Fix CFLAGS and hopefully #430594.
Original commit message from CVS:
* gst/app/Makefile.am:
Fix CFLAGS and hopefully #430594.
Sebastian Dröge [Tue, 17 Apr 2007 02:53:16 +0000 (02:53 +0000)]
gst-libs/gst/riff/riff-media.c: Allow random depths between 1 and 32 instead of only multiplies of 8.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Allow random depths between 1 and 32 instead of only multiplies of 8.
Sebastian Dröge [Tue, 17 Apr 2007 02:04:21 +0000 (02:04 +0000)]
gst-libs/gst/riff/riff-media.c: Set the maximum number of channels for PCM and float in the correct place to have it ...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Set the maximum number of channels for PCM and float in the correct
place to have it also used when creating the template caps.
Sebastian Dröge [Tue, 17 Apr 2007 01:56:07 +0000 (01:56 +0000)]
gst-libs/gst/riff/riff-media.c: Correctly support 4, 6 and 8 channels with normal PCM and float wav files.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Correctly support 4, 6 and 8 channels with normal PCM and float
wav files.
Fix the depth and signedness calculation in extensible wav files and
also handle 1, 2, 4, 6, 8 channels here when a file without channel
mask is found.
Add support for float, alaw and mulaw in extensible wav files.
This allows correct playback of all but 5 files from
http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples.html
(gst_riff_create_audio_template_caps):
Add voxware and float formats to the template caps.
Vincent Torri [Mon, 16 Apr 2007 22:20:03 +0000 (22:20 +0000)]
ext/pango/gstclockoverlay.c: Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
Original commit message from CVS:
Patch by: Vincent Torri <vtorri at univ-evry dot fr>
* ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
* gst/audioresample/gstaudioresample.c: (audioresample_do_output):
Use the correct format strings for integer formats.
Thomas Vander Stichele [Mon, 16 Apr 2007 21:44:34 +0000 (21:44 +0000)]
fix release date
Original commit message from CVS:
fix release date
Thomas Vander Stichele [Mon, 16 Apr 2007 21:42:13 +0000 (21:42 +0000)]
fix release date
Original commit message from CVS:
fix release date
Thomas Vander Stichele [Sun, 15 Apr 2007 14:35:53 +0000 (14:35 +0000)]
ext/theora/theoraenc.c (theora_buffer_from_packet, theora_enc_chain): Don't use pad_alloc_buffer_and_set_caps to crea...
Original commit message from CVS:
* ext/theora/theoraenc.c (theora_buffer_from_packet, theora_enc_chain):
Don't use pad_alloc_buffer_and_set_caps to create a small header
packet, or, worse, to create a big temporary video buffer using the
src pad.
Thomas Vander Stichele [Sat, 14 Apr 2007 12:34:55 +0000 (12:34 +0000)]
gst/gdp/gstgdppay.c (gst_gdp_pay_chain): tests/check/pipelines/streamheader.c (tag_event_probe_cb,
Original commit message from CVS:
* gst/gdp/gstgdppay.c (gst_gdp_pay_chain):
* tests/check/pipelines/streamheader.c (tag_event_probe_cb,
GST_START_TEST, buffer_probe_cb, GST_START_TEST):
Fix a bug where serialized IN_CAPS buffers needed to be set IN_CAPS.
Thomas Vander Stichele [Fri, 13 Apr 2007 22:10:58 +0000 (22:10 +0000)]
add debug
Original commit message from CVS:
add debug
Thomas Vander Stichele [Fri, 13 Apr 2007 21:55:31 +0000 (21:55 +0000)]
tests/check/pipelines/streamheader.c (tag_event_probe_cb,
Original commit message from CVS:
* tests/check/pipelines/streamheader.c (tag_event_probe_cb,
GST_START_TEST, n_in_caps, buffer_probe_cb, GST_START_TEST,
streamheader_suite):
Add another test set up for failure
Thomas Vander Stichele [Fri, 13 Apr 2007 21:09:04 +0000 (21:09 +0000)]
debug changes
Original commit message from CVS:
debug changes
Thomas Vander Stichele [Fri, 13 Apr 2007 21:08:11 +0000 (21:08 +0000)]
tests/check/Makefile.am: tests/check/pipelines/streamheader.c (n_tags, tag_event_probe_cb,
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/pipelines/streamheader.c (n_tags, tag_event_probe_cb,
GST_START_TEST, streamheader_suite, main):
Add a test for the streamheader bug Wim fixed.
Jan Schmidt [Fri, 13 Apr 2007 11:42:34 +0000 (11:42 +0000)]
ext/theora/theoradec.c: Fix misleading comment.
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_dec_sink_event):
Fix misleading comment.
Stefan Kost [Fri, 13 Apr 2007 06:17:45 +0000 (06:17 +0000)]
gst-libs/gst/riff/riff-media.c: More sanity checks for the header fields.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
More sanity checks for the header fields.
Tim-Philipp Müller [Thu, 12 Apr 2007 16:36:36 +0000 (16:36 +0000)]
gst-libs/gst/tag/tags.c: Try encodings from all environment variables, not just those in the first environment variab...
Original commit message from CVS:
* gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
Try encodings from all environment variables, not just those in the
first environment variable that is set.
Wim Taymans [Thu, 12 Apr 2007 15:00:03 +0000 (15:00 +0000)]
gst/videorate/gstvideorate.c: Add some debug.
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
(gst_video_rate_chain):
Add some debug.
* tests/check/elements/videorate.c: (GST_START_TEST),
(videorate_suite):
Added check for videorate changing caps handling. Closes #421834.
Michael Smith [Thu, 12 Apr 2007 12:57:33 +0000 (12:57 +0000)]
ext/vorbis/vorbisdec.c: Use scale functions to avoid overflow when calculating duration of vorbis buffers.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
Use scale functions to avoid overflow when calculating duration of
vorbis buffers.
Tim-Philipp Müller [Thu, 12 Apr 2007 12:19:20 +0000 (12:19 +0000)]
API: add gst_tag_freeform_string_to_utf8() (#405072).
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
API: add gst_tag_freeform_string_to_utf8() (#405072).
* gst-libs/gst/tag/gstid3tag.c: (gst_tag_extract_id3v1_string):
Use gst_tag_freeform_string_to_utf8() here.
Thomas Vander Stichele [Thu, 12 Apr 2007 10:38:03 +0000 (10:38 +0000)]
log tweaking
Original commit message from CVS:
log tweaking
Wim Taymans [Thu, 12 Apr 2007 10:03:22 +0000 (10:03 +0000)]
gst/gdp/gstgdppay.c: Make sure we set the IN_CAPS flag correctly.
Original commit message from CVS:
* gst/gdp/gstgdppay.c: (gst_gdp_pay_chain),
(gst_gdp_pay_sink_event):
Make sure we set the IN_CAPS flag correctly.
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
Get the IN_CAPS flag before we call functions that mess with the flags.
Thomas Vander Stichele [Tue, 10 Apr 2007 20:37:05 +0000 (20:37 +0000)]
gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader, gst_gdp_pay_chain, gst_gdp_pay_sink_event):
Original commit message from CVS:
* gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader,
gst_gdp_pay_chain, gst_gdp_pay_sink_event):
Only stamp buffers with offset/offset_end right before they get
pushed. This ensures offset continuity, which was not the case
before as shown by
gst-launch -v -m audiotestsrc num-buffers=10 ! audioconvert ! vorbisenc ! gdppay ! identity check-imperfect-offset=TRUE ! fakesink silent=TRUE
Thomas Vander Stichele [Tue, 10 Apr 2007 20:25:06 +0000 (20:25 +0000)]
adding debugging
Original commit message from CVS:
adding debugging
Christian Schaller [Tue, 10 Apr 2007 11:23:18 +0000 (11:23 +0000)]
update spec file for RTP changes
Original commit message from CVS:
update spec file for RTP changes
Wim Taymans [Fri, 6 Apr 2007 12:58:06 +0000 (12:58 +0000)]
gst/playback/gstplaybin.c: Activate sync in playbin, we are ready to handle it for live streams.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (add_sink),
(gst_play_bin_change_state):
Activate sync in playbin, we are ready to handle it for live streams.
Tim-Philipp Müller [Fri, 6 Apr 2007 09:56:18 +0000 (09:56 +0000)]
tests/check/elements/playbin.c: Add small test for stream-info-value-array code paths.
Original commit message from CVS:
* tests/check/elements/playbin.c:
(test_sink_usage_video_only_stream), (playbin_suite):
Add small test for stream-info-value-array code paths.
Wim Taymans [Thu, 5 Apr 2007 15:44:40 +0000 (15:44 +0000)]
gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to create invalid calibration parameters by making the internal time...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving):
Don't try to create invalid calibration parameters by making the
internal time go backwards, instead make external time go forward.
Tommi Myöhänen [Thu, 5 Apr 2007 10:27:06 +0000 (10:27 +0000)]
gst/playback/gstplaybasebin.c: Fix leak in add_stream(), when g_value_set_object() increases the refcount of streamin...
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/playback/gstplaybasebin.c: (add_stream):
Fix leak in add_stream(), when g_value_set_object() increases the
refcount of streaminfo object. Fixes #426250.
David Schleef [Wed, 4 Apr 2007 02:45:03 +0000 (02:45 +0000)]
gst/videotestsrc/: Add a test pattern called "circular", which has concentric rings with varying radial frequency. T...
Original commit message from CVS:
* gst/videotestsrc/gstvideotestsrc.c:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
Add a test pattern called "circular", which has concentric
rings with varying radial frequency. The main purpose of this
pattern is to test fidelity loss in a filter or scaler element.
Notably, this pattern is scale invariant, and is optimally viewed
with a width (and height) of 400.
Tommi Myöhänen [Tue, 3 Apr 2007 11:10:52 +0000 (11:10 +0000)]
gst/playback/gstdecodebin2.c: Decodebin2 doesn't unref pads it obtains in some occasions:
Original commit message from CVS:
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
* gst/playback/gstdecodebin2.c: (connect_pad), (expose_pad),
(deactivate_free_recursive):
Decodebin2 doesn't unref pads it obtains in some occasions:
- multiqueue src pads, when either connecting further or exposing
- sink pads of new autoplugged elements
- peer pads when recursively freeing elements
Fixes #425455.
Sebastian Dröge [Fri, 30 Mar 2007 17:05:23 +0000 (17:05 +0000)]
gst-libs/gst/riff/riff-media.c: Add audio/x-raw-float support, now that audioconvert support non-native endianness fl...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Add audio/x-raw-float support, now that audioconvert support
non-native endianness floats.
Tim-Philipp Müller [Fri, 30 Mar 2007 15:00:49 +0000 (15:00 +0000)]
docs/libs/gst-plugins-base-libs-docs.sgml: gstreamer-plugins-base.pc doesn't exist, it's gstreamer-plugins-base-0.10.pc.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
gstreamer-plugins-base.pc doesn't exist, it's
gstreamer-plugins-base-0.10.pc.
René Stadler [Thu, 29 Mar 2007 18:42:34 +0000 (18:42 +0000)]
with some minor changes
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler dot de>
with some minor changes
* gst-libs/gst/floatcast/floatcast.h:
Use more efficient float endianness conversion functions that don't
involve 2 function calls per value.
* gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
(check_default), (audio_convert_prepare_context):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps), (make_lossless_changes):
Support non-native endianness floats as input and output.
Fixes #339838.
* tests/check/elements/audioconvert.c: (verify_convert),
(GST_START_TEST):
Add unit tests for the non-native endianness float conversions.
Wim Taymans [Thu, 29 Mar 2007 16:23:53 +0000 (16:23 +0000)]
gst-libs/gst/rtp/gstbasertpdepayload.*: Add Private structure.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_base_init),
(gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_set_gst_timestamp),
(gst_base_rtp_depayload_change_state),
(gst_base_rtp_depayload_set_property),
(gst_base_rtp_depayload_get_property):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Add Private structure.
Bring element code to 2007.
Parse clock-base caps param and use it when generating the
newsegment.
Reset variables before going to PAUSED.
Fix some docs.
Wim Taymans [Thu, 29 Mar 2007 16:20:31 +0000 (16:20 +0000)]
Add RTCP docs.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_get_adapter):
Add RTCP docs.
Fix some more docs.
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtp/gstrtcpbuffer.c:
(gst_rtcp_buffer_new_take_data), (gst_rtcp_buffer_new_copy_data),
(gst_rtcp_buffer_validate_data), (gst_rtcp_buffer_validate),
(gst_rtcp_buffer_get_packet_count), (read_packet_header),
(gst_rtcp_buffer_get_first_packet), (gst_rtcp_packet_move_to_next),
(gst_rtcp_buffer_add_packet), (gst_rtcp_packet_remove),
(gst_rtcp_packet_get_padding), (gst_rtcp_packet_get_type),
(gst_rtcp_packet_get_count), (gst_rtcp_packet_get_length),
(gst_rtcp_packet_sr_get_sender_info),
(gst_rtcp_packet_sr_set_sender_info),
(gst_rtcp_packet_rr_get_ssrc), (gst_rtcp_packet_rr_set_ssrc),
(gst_rtcp_packet_get_rb_count), (gst_rtcp_packet_get_rb),
(gst_rtcp_packet_add_rb), (gst_rtcp_packet_set_rb),
(gst_rtcp_packet_sdes_get_chunk_count),
(gst_rtcp_packet_sdes_first_chunk),
(gst_rtcp_packet_sdes_next_chunk), (gst_rtcp_packet_sdes_get_ssrc),
(gst_rtcp_packet_sdes_first_item),
(gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_item),
(gst_rtcp_packet_bye_get_ssrc_count),
(gst_rtcp_packet_bye_get_nth_ssrc), (gst_rtcp_packet_bye_add_ssrc),
(gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
(gst_rtcp_packet_bye_get_reason_len),
(gst_rtcp_packet_bye_get_reason), (gst_rtcp_packet_bye_set_reason):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Add new helper object for parsing and creating RTCP messages.
Sebastian Dröge [Thu, 29 Mar 2007 12:07:02 +0000 (12:07 +0000)]
gst-libs/gst/riff/riff-media.c: PCM samples with width=8 must be always unsigned, no matter what depth they have.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
PCM samples with width=8 must be always unsigned, no matter what
depth they have.
Andy Wingo [Thu, 29 Mar 2007 11:24:47 +0000 (11:24 +0000)]
gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make perfect offsets also, not just timestamps.
Original commit message from CVS:
2007-03-29 Andy Wingo <wingo@pobox.com>
* gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make
perfect offsets also, not just timestamps.
* tests/check/elements/videorate.c (test_more): Test that given
any incoming offsets, that videorate produces perfect offsets.
Wim Taymans [Thu, 29 Mar 2007 10:19:45 +0000 (10:19 +0000)]
gst-libs/gst/riff/riff-ids.h: Add some more RIFF formats.
Original commit message from CVS:
* gst-libs/gst/riff/riff-ids.h:
Add some more RIFF formats.
Wim Taymans [Thu, 29 Mar 2007 10:17:52 +0000 (10:17 +0000)]
gst-libs/gst/rtp/gstrtpbuffer.*: Fix fixed payload names and docs.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_default_clock_rate):
* gst-libs/gst/rtp/gstrtpbuffer.h:
Fix fixed payload names and docs.
Added method to get the default clock rates of fixed payload types.
API: GstRTPBuffer::gst_rtp_buffer_default_clock_rate()
Zaheer Abbas Merali [Wed, 28 Mar 2007 15:24:40 +0000 (15:24 +0000)]
tests/check/pipelines/.cvsignore: Add new vorbisdec test to cvsignore.
Original commit message from CVS:
* tests/check/pipelines/.cvsignore:
Add new vorbisdec test to cvsignore.
Wim Taymans [Wed, 28 Mar 2007 14:50:47 +0000 (14:50 +0000)]
gst-libs/gst/audio/gstbaseaudiosink.*: Store private stuff in GstBaseAudioSinkPrivate.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
(gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
(gst_base_audio_sink_query), (gst_base_audio_sink_get_time),
(gst_base_audio_sink_set_property),
(gst_base_audio_sink_get_property), (gst_base_audio_sink_event),
(clock_convert_external), (gst_base_audio_sink_resample_slaving),
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_handle_slaving), (gst_base_audio_sink_render),
(gst_base_audio_sink_async_play):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Store private stuff in GstBaseAudioSinkPrivate.
Add configurable clock slaving modes property.
API:: GstBaseAudioSink::slave-method property
Some more latency reporting tweaks.
Added skew based clock slaving correction and make it the default until
the resampling method is more robust.
Sebastian Dröge [Tue, 27 Mar 2007 12:44:14 +0000 (12:44 +0000)]
gst/audioconvert/audioconvert.c: Add docs to the integer pack functions and implement proper rounding. Before we had ...
Original commit message from CVS:
* gst/audioconvert/audioconvert.c:
Add docs to the integer pack functions and implement proper
rounding. Before we had rounding towards negative infinity, i.e.
always the smaller number was taken. Now we use natural rounding,
i.e. rounding to the nearest integer and to the one with the largest
absolute value for X.5. The old rounding introduced some minor
distortions. Fixes #420079
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Fix one unit test that assumed the old rounding and added unit tests
for checking signed/unsigned int16 <-> signed/unsigned int16 with
depth 8, one for signed int16 <-> unsigned int16 and one for the new
rounding from signed int32 to signed/unsigned int16.
Michael Smith [Tue, 27 Mar 2007 11:31:17 +0000 (11:31 +0000)]
gst/audioconvert/gstaudioconvert.c: Fix typo in debug line introduced recently, as pointed out on irc.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (strip_width_64),
(gst_audio_convert_transform_caps):
Fix typo in debug line introduced recently, as pointed out on irc.
Tim-Philipp Müller [Tue, 27 Mar 2007 10:17:16 +0000 (10:17 +0000)]
Make sure we parse floating-point numbers in vorbis comments correctly with either '.' or ',' as separator, no matter...
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
* tests/check/libs/tag.c: (GST_START_TEST):
Make sure we parse floating-point numbers in vorbis comments
correctly with either '.' or ',' as separator, no matter what
the current locale is. Add unit test for this too.
Thomas Vander Stichele [Tue, 27 Mar 2007 09:37:42 +0000 (09:37 +0000)]
commit new file
Original commit message from CVS:
commit new file
René Stadler [Mon, 26 Mar 2007 22:38:19 +0000 (22:38 +0000)]
gst-libs/gst/tag/gstvorbistag.c: When writing out floating-point numbers to vorbis comment tags, always use the same ...
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler de>
* gst-libs/gst/tag/gstvorbistag.c: (gst_tag_to_vorbis_comments):
When writing out floating-point numbers to vorbis comment tags, always
use the same character as separator no matter what the current locale is
(fixes #423051).
* tests/check/libs/tag.c: (GST_START_TEST):
Add unit tests for replaygain tags in vorbis comments (closes #423055).
Thomas Vander Stichele [Mon, 26 Mar 2007 20:56:35 +0000 (20:56 +0000)]
ext/vorbis/vorbisdec.c (vorbis_dec_push_forward, vorbis_handle_data_packet):
Original commit message from CVS:
* ext/vorbis/vorbisdec.c (vorbis_dec_push_forward,
vorbis_handle_data_packet):
Correctly set DURATION to generate a timestamp-continuous stream.
One bug left at the end; see
ihttp://bugzilla.gnome.org/show_bug.cgi?id=423086
* tests/check/Makefile.am:
* tests/check/pipelines/vorbisenc.c (GST_START_TEST):
Add a test to check this. Without the above patch this test fails.
Jan Schmidt [Mon, 26 Mar 2007 11:44:07 +0000 (11:44 +0000)]
gst-libs/gst/rtp/Makefile.am: The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
Original commit message from CVS:
* gst-libs/gst/rtp/Makefile.am:
The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
Christian Schaller [Fri, 23 Mar 2007 15:43:24 +0000 (15:43 +0000)]
update spec file
Original commit message from CVS:
update spec file
Michael Smith [Fri, 23 Mar 2007 12:32:33 +0000 (12:32 +0000)]
gst/videorate/gstvideorate.c: If videorate changes caps, we can no longer use the old buffer (which may have a differ...
Original commit message from CVS:
* gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
(gst_video_rate_reset), (gst_video_rate_chain):
If videorate changes caps, we can no longer use the old buffer
(which may have a different size, incompatible with our caps).
So don't do that; just duplicate the new frame more times.
Jan Schmidt [Thu, 22 Mar 2007 17:43:52 +0000 (17:43 +0000)]
gst/playback/gstplaybin.c: Remove playbin's override of the set_clock vmethod. It's irrelevant after Wim's commit on ...
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
Remove playbin's override of the set_clock vmethod. It's irrelevant
after Wim's commit on the 19th.
Thomas Vander Stichele [Thu, 22 Mar 2007 14:37:08 +0000 (14:37 +0000)]
gst-libs/gst/app/Makefile.am: Use GST_ALL_LDFLAGS, which actually exists, but maybe David can confirm that was what h...
Original commit message from CVS:
* gst-libs/gst/app/Makefile.am:
Use GST_ALL_LDFLAGS, which actually exists, but maybe David
can confirm that was what he wanted.
Wim Taymans [Thu, 22 Mar 2007 09:26:02 +0000 (09:26 +0000)]
ext/gnomevfs/gstgnomevfssrc.*: Don't cache file sizes. Fixes #341078.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_get_size),
(gst_gnome_vfs_src_start), (gst_gnome_vfs_src_stop):
* ext/gnomevfs/gstgnomevfssrc.h:
Don't cache file sizes. Fixes #341078.
Tim-Philipp Müller [Wed, 21 Mar 2007 11:03:23 +0000 (11:03 +0000)]
gst/playback/gstplaybin.c: Use GST_PTR_FORMAT to log caps.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (add_sink):
Use GST_PTR_FORMAT to log caps.
Young-Ho Cha [Wed, 21 Mar 2007 10:23:11 +0000 (10:23 +0000)]
gst/subparse/samiparse.c: Special-case some more colour names that pango doesn't handle by default. Fixes #420578.
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian net>
* gst/subparse/samiparse.c: (handle_start_font):
Special-case some more colour names that pango doesn't handle by
default. Fixes #420578.
Michael Smith [Tue, 20 Mar 2007 11:49:55 +0000 (11:49 +0000)]
ext/vorbis/vorbisenc.c: If we get a zero-sized input buffer, don't pass it to libvorbis, as that marks EOS internally...
Original commit message from CVS:
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
If we get a zero-sized input buffer, don't pass it to libvorbis, as
that marks EOS internally. After that, libvorbis will buffer all
input data, and encode none of it, eventually leading to memory
exhaustion.
Wim Taymans [Mon, 19 Mar 2007 10:52:50 +0000 (10:52 +0000)]
gst/playback/gstdecodebin.c: Don't post STATE_DIRTY anymore.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (remove_fakesink):
Don't post STATE_DIRTY anymore.
* gst/playback/gstplaybin.c: (add_sink), (gst_play_bin_send_event),
(gst_play_bin_change_state):
Remove stream_time reset in seek handling, core does that now.
Disable clocking for live pipelines by forcing a NULL clock to the
complete pipeline, core is too smart now for our previous hack.
We can always autoplug in PAUSED now.
David Schleef [Sun, 18 Mar 2007 03:14:01 +0000 (03:14 +0000)]
REQUIREMENTS: Update this file, change the formatting to make it more consistent, plus more machine readable.
Original commit message from CVS:
* REQUIREMENTS: Update this file, change the formatting to make
it more consistent, plus more machine readable.
Michael Smith [Fri, 16 Mar 2007 17:29:09 +0000 (17:29 +0000)]
gst/audioconvert/gstaudioconvert.c: Previous fix was too simplistic, and broke the tests. Use a better approach; only...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
(strip_width_64), (append_with_other_format):
Previous fix was too simplistic, and broke the tests. Use a better
approach; only strip 64 from widths for integer audio.
Michael Smith [Fri, 16 Mar 2007 16:42:23 +0000 (16:42 +0000)]
gst/audioconvert/gstaudioconvert.c: We don't support 64 bit integer audio, so don't try to claim we can.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
(gst_audio_convert_transform_caps):
We don't support 64 bit integer audio, so don't try to claim we can.
Stops us producing caps don't match our template caps.
Update comments.
Michael Smith [Thu, 15 Mar 2007 10:52:21 +0000 (10:52 +0000)]
gst/audioresample/gstaudioresample.c: Don't trigger discontinuities for very small imperfections; a filter flush will...
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c:
(audioresample_check_discont), (audioresample_transform):
Don't trigger discontinuities for very small imperfections; a filter
flush will sound bad, and many plugins have rounding errors leading
to these.
Philippe Kalaf [Wed, 14 Mar 2007 21:11:18 +0000 (21:11 +0000)]
gst-libs/gst/rtp/gstbasertpaudiopayload.*: olivier.crete@collabora.co.uk.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
Add min-ptime property to RTP base audio payloader. Patch by
olivier.crete@collabora.co.uk.
Fixes #415001
Indentation/whitespace/documentation fixes.
Julien Moutte [Wed, 14 Mar 2007 17:16:30 +0000 (17:16 +0000)]
gst/audioresample/gstaudioresample.c: Handle discontinuous streams.
Original commit message from CVS:
2007-03-14 Julien MOUTTE <julien@moutte.net>
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
(audioresample_transform_size), (audioresample_do_output),
(audioresample_transform), (audioresample_pushthrough): Handle
discontinuous streams.
* gst/audioresample/gstaudioresample.h:
* tests/check/elements/audioresample.c:
(test_discont_stream_instance), (GST_START_TEST),
(audioresample_suite): Add a test for discontinuous streams.
* win32/common/config.h: Updated.
Thomas Vander Stichele [Wed, 14 Mar 2007 15:16:23 +0000 (15:16 +0000)]
po/: Update translations from translation project.
Original commit message from CVS:
* po/af.po:
* po/az.po:
* po/cs.po:
* po/en_GB.po:
* po/it.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
Update translations from translation project.
Thomas Vander Stichele [Wed, 14 Mar 2007 15:05:32 +0000 (15:05 +0000)]
add buffer logging
Original commit message from CVS:
add buffer logging
Thomas Vander Stichele [Wed, 14 Mar 2007 14:48:12 +0000 (14:48 +0000)]
gst/audioresample/: Since I really am not interested in a debug line for each sample being processed, move the librar...
Original commit message from CVS:
* gst/audioresample/debug.h:
* gst/audioresample/resample.c: (resample_init):
Since I really am not interested in a debug line for each sample
being processed, move the library's debugging to its own category,
libaudioresample
Thomas Vander Stichele [Wed, 14 Mar 2007 14:09:21 +0000 (14:09 +0000)]
add debugging and reformat docs
Original commit message from CVS:
add debugging and reformat docs
Michael Smith [Mon, 12 Mar 2007 23:29:07 +0000 (23:29 +0000)]
ext/theora/theoradec.c: Since the plugin doesn't support anything other than 4:2:0 right now, post an error and fail ...
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_handle_type_packet):
Since the plugin doesn't support anything other than 4:2:0 right
now, post an error and fail if we get something else. Won't matter
until libtheora supports the other pixel formats, but hopefully
that'll be soon...
Alex Lancaster [Mon, 12 Mar 2007 15:50:35 +0000 (15:50 +0000)]
I'm too lazy to comment this
Original commit message from CVS:
Mention Patch by: Alex Lancaster in a recent commit.
Jan Schmidt [Mon, 12 Mar 2007 11:47:42 +0000 (11:47 +0000)]
examples/app/.cvsignore: The buildbot demands .cvsignore files, and I comply.
Original commit message from CVS:
* examples/app/.cvsignore:
The buildbot demands .cvsignore files, and I comply.
David Schleef [Sun, 11 Mar 2007 00:48:26 +0000 (00:48 +0000)]
Add appsrc/appsink example.
Original commit message from CVS:
* configure.ac:
* examples/Makefile.am:
* examples/app/Makefile.am:
* examples/app/appsrc_ex.c:
Add appsrc/appsink example.
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/app/gstapp.c:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsink.h:
* gst/app/gstapp.c:
Add appsink.
Sébastien Moutte [Sat, 10 Mar 2007 15:59:33 +0000 (15:59 +0000)]
gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_guint64_to_gdouble for conversion.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:(gst_base_audio_sink_render):
Use gst_guint64_to_gdouble for conversion.
* win32/MANIFEST:
Add new files to the win32 MANIFEST.
* win32/common/libgstaudio.def:
* win32/common/libgstpbutils.def:
Add new exported functions.
* win32/vs6/gst_plugins_base.dsw:
* win32/vs6/libgstdecodebin.dsp:
* win32/vs6/libgstplaybin.dsp:
Change the link to libgstpbutils.lib.
* win32/vs6/libgstdecodebin2.dsp:
Add a new project for decodebin2.
* win32/vs6/libgstpbutils.dsp:
Add a new project for pbutils.
Tim-Philipp Müller [Sat, 10 Mar 2007 12:18:58 +0000 (12:18 +0000)]
gst-libs/gst/tag/gstvorbistag.c: Also accept partial dates with only year and month, like 1999-12-00 (fixes #410396 e...
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
Also accept partial dates with only year and month,
like 1999-12-00 (fixes #410396 even more).
* tests/check/libs/tag.c: (GST_START_TEST):
Add unit test for the above.
Tim-Philipp Müller [Sat, 10 Mar 2007 11:21:08 +0000 (11:21 +0000)]
tests/check/elements/subparse.c: Add unit test for MPL2 subtitle format (#413799).
Original commit message from CVS:
* tests/check/elements/subparse.c: (GST_START_TEST),
(subparse_suite):
Add unit test for MPL2 subtitle format (#413799).
Kamil Pawlowski [Sat, 10 Mar 2007 11:17:52 +0000 (11:17 +0000)]
gst/subparse/: Add support for MPL2 subtitle format (#413799).
Original commit message from CVS:
Patch by: Kamil Pawlowski <kamilpe gmail com>
* gst/subparse/Makefile.am:
* gst/subparse/gstsubparse.c:
(gst_sub_parse_data_format_autodetect),
(gst_sub_parse_format_autodetect), (gst_sub_parse_sink_event),
(gst_subparse_type_find):
* gst/subparse/gstsubparse.h:
* gst/subparse/mpl2parse.c: (mpl2_parse_line), (parse_mpl2):
* gst/subparse/mpl2parse.h:
Add support for MPL2 subtitle format (#413799).
Tim-Philipp Müller [Fri, 9 Mar 2007 17:33:17 +0000 (17:33 +0000)]
configure.ac: We require core CVS for the new buffer metadata copy functions.
Original commit message from CVS:
* configure.ac:
We require core CVS for the new buffer metadata copy functions.
Wim Taymans [Fri, 9 Mar 2007 16:51:13 +0000 (16:51 +0000)]
gst-libs/gst/tag/gstid3tag.c: Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
Original commit message from CVS:
* gst-libs/gst/tag/gstid3tag.c:
Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
Fixes #414496.
Wim Taymans [Fri, 9 Mar 2007 16:46:35 +0000 (16:46 +0000)]
ext/libvisual/visual.c: Improve adapter usage and comments.
Original commit message from CVS:
* ext/libvisual/visual.c: (gst_visual_sink_setcaps),
(gst_vis_src_negotiate), (get_buffer), (gst_visual_chain):
Improve adapter usage and comments.
Wim Taymans [Fri, 9 Mar 2007 16:38:06 +0000 (16:38 +0000)]
Use new metadata copy function.
Original commit message from CVS:
* ext/pango/gsttextrender.c: (gst_text_render_chain):
* ext/vorbis/vorbistag.c: (gst_vorbis_tag_parse_packet):
* gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netbuffer_copy):
Use new metadata copy function.
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_transform):
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
Basetransform copied the metadata for us.
Tim-Philipp Müller [Fri, 9 Mar 2007 16:28:04 +0000 (16:28 +0000)]
ext/pango/gsttextoverlay.c: Some more logging. Only accept newsegment events in TIME format and send a WARNING messag...
Original commit message from CVS:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event),
(gst_text_overlay_video_event):
Some more logging. Only accept newsegment events in TIME format and
send a WARNING message if they are not in TIME format.
* gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
(gst_sub_parse_init), (gst_sub_parse_src_event), (handle_buffer),
(gst_sub_parse_chain), (gst_sub_parse_sink_event):
* gst/subparse/gstsubparse.h:
No need to allocate GstSegment structure dynamically, just put it
into the instance structure; ignore newsegment events in BYTE
format and in particular don't let it overwrite our saved TIME
segment from the last seek.
projects / platform / upstream / gstreamer.git / log