Wim Taymans [Thu, 17 May 2007 16:27:32 +0000 (16:27 +0000)]
gst/playback/gstdecodebin2.c: Make decodebin2 autoplug depayloaders too.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter):
Make decodebin2 autoplug depayloaders too.
* gst/playback/gsturidecodebin.c: (source_new_pad):
Set the newly created decoder in a usable state when autoplugging a
dynamic source such as RTSP.
Tim-Philipp Müller [Thu, 17 May 2007 16:11:03 +0000 (16:11 +0000)]
gst/playback/gststreaminfo.c: Ignore video-codec tag for audio streams and ignore audio-codec tags for video streams....
Original commit message from CVS:
* gst/playback/gststreaminfo.c: (cb_probe):
Ignore video-codec tag for audio streams and ignore audio-codec tags
for video streams. Should make codec name collection a bit more
robust against sloppy demuxers that send tag events containing both
tags down each pad.
Wim Taymans [Thu, 17 May 2007 15:22:44 +0000 (15:22 +0000)]
gst/playback/gstqueue2.c: Tweak the buffering thresholds a little.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (update_rates):
Tweak the buffering thresholds a little.
Update the buffer size with the previously calculate rate instead of
only when we calculate a new rate so that we get smoother buffering
updates.
* gst/playback/Makefile.am:
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_base_init),
(gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
(gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
(gst_uri_decode_bin_get_property), (unknown_type),
(add_element_stream), (no_more_pads_full), (no_more_pads),
(source_no_more_pads), (new_decoded_pad), (array_has_value),
(gen_source_element), (has_all_raw_caps), (analyse_source),
(remove_decoders), (make_decoder), (remove_source),
(source_new_pad), (setup_source), (decoder_query_init),
(decoder_query_duration_fold), (decoder_query_duration_done),
(decoder_query_position_fold), (decoder_query_position_done),
(decoder_query_latency_fold), (decoder_query_latency_done),
(decoder_query_seeking_fold), (decoder_query_seeking_done),
(decoder_query_generic_fold), (gst_uri_decode_bin_query),
(gst_uri_decode_bin_change_state), (plugin_init):
New element that intergrates a source, optional buffering element and
decodebin.
Tim-Philipp Müller [Thu, 17 May 2007 14:17:17 +0000 (14:17 +0000)]
configure.ac: Bump libtheora requirement to 1.0alpha5 for the pixformat check (also has a .pc file, so we don't need ...
Original commit message from CVS:
* configure.ac:
Bump libtheora requirement to 1.0alpha5 for the pixformat check
(also has a .pc file, so we don't need the fallback check any
longer). Fixes #438840.
Wim Taymans [Thu, 17 May 2007 13:36:11 +0000 (13:36 +0000)]
gst/playback/gstqueue2.c: fix build.
Original commit message from CVS:
* gst/playback/gstqueue2.c: (gst_queue_get_type),
(gst_queue_class_init), (gst_queue_finalize), (update_time_level),
(apply_segment), (apply_buffer), (update_buffering),
(reset_rate_timer), (update_rates), (gst_queue_locked_flush),
(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
(gst_queue_handle_sink_event), (gst_queue_is_filled),
(gst_queue_chain), (gst_queue_push_one), (gst_queue_loop),
(plugin_init):
fix build.
Wim Taymans [Thu, 17 May 2007 11:57:44 +0000 (11:57 +0000)]
gst/playback/: On our way to playbin2 this is the new network queue that does buffering all by itself using high and ...
Original commit message from CVS:
* gst/playback/Makefile.am:
* gst/playback/gstqueue2.c: (gst_queue_get_type),
(gst_queue_class_init), (gst_queue_init), (gst_queue_finalize),
(gst_queue_getcaps), (gst_queue_bufferalloc),
(gst_queue_acceptcaps), (update_time_level), (apply_segment),
(apply_buffer), (update_buffering), (reset_rate_timer),
(update_rates), (gst_queue_locked_flush),
(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
(gst_queue_handle_sink_event), (gst_queue_is_empty),
(gst_queue_is_filled), (gst_queue_chain), (gst_queue_push_one),
(gst_queue_loop), (gst_queue_handle_src_event),
(gst_queue_handle_src_query), (gst_queue_sink_activate_push),
(gst_queue_src_activate_push), (gst_queue_change_state),
(gst_queue_set_property), (gst_queue_get_property), (plugin_init):
On our way to playbin2 this is the new network queue that does buffering
all by itself using high and low watermarks. It can also measure up and
downstream bandwidth to optimally size the queue.
Michael Smith [Thu, 17 May 2007 11:16:14 +0000 (11:16 +0000)]
gst/: Use the segment->last_stop value to calculate the next timestamp to generate after a seek; not the segment->sta...
Original commit message from CVS:
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
* gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_do_seek):
Use the segment->last_stop value to calculate the next timestamp to
generate after a seek; not the segment->start value.
David Schleef [Tue, 15 May 2007 20:14:06 +0000 (20:14 +0000)]
docs/Makefile.am: Install docs even when --disable-gtk-doc is disabled. This matches the behavior of gtk+. Fixes #3...
Original commit message from CVS:
* docs/Makefile.am: Install docs even when --disable-gtk-doc
is disabled. This matches the behavior of gtk+. Fixes #349099.
Wim Taymans [Tue, 15 May 2007 17:11:09 +0000 (17:11 +0000)]
ext/ogg/gstoggdemux.c: Some more chained streaming ogg timestamp fixes.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
(gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
Some more chained streaming ogg timestamp fixes.
Wim Taymans [Tue, 15 May 2007 16:46:10 +0000 (16:46 +0000)]
ext/ogg/gstoggdemux.c: Add some FIXMEs.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
(gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
(gst_ogg_demux_handle_page):
Add some FIXMEs.
Fix chain start/stop segment handling based on patch by
<ahalda at cs dot mcgill dot ca> see #320984.
Michael Smith [Tue, 15 May 2007 15:33:54 +0000 (15:33 +0000)]
configure.ac: We don't require a C++ compiler. So don't require one.
Original commit message from CVS:
* configure.ac:
We don't require a C++ compiler. So don't require one.
Stefan Kost [Tue, 15 May 2007 15:29:17 +0000 (15:29 +0000)]
ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds, gst_alsa_mixer_check, gst_alsa_mixer_dispatch, gst_alsa_mixer_...
Original commit message from CVS:
* ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds,
gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
gst_alsa_mixer_finalize, gst_alsa_mixer_handle_source_callback,
gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_update_option,
gst_alsa_mixer_update_track):
Apply some of the cleanup Tim suggested in #152864 afterwards.
Marc-Andre Lureau [Tue, 15 May 2007 14:01:26 +0000 (14:01 +0000)]
ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch, _GstAlsaMixerWatch, source, n_poll_fds, poll_fds, gst_alsa_...
Original commit message from CVS:
patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
* ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch,
_GstAlsaMixerWatch, source, n_poll_fds, poll_fds,
gst_alsa_mixer_watch_funcs, gst_alsa_mixer_prepare,
gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
gst_alsa_mixer_finalize, gst_alsa_mixer_find_master_mixer,
gst_alsa_mixer_handle_source_callback,
gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_free,
gst_alsa_mixer_get_volume, gst_alsa_mixer_set_volume,
gst_alsa_mixer_set_mute, gst_alsa_mixer_set_record,
gst_alsa_mixer_get_option, gst_alsa_mixer_update_option,
gst_alsa_mixer_update_track, _gst_alsa_mixer_set_interface):
* ext/alsa/gstalsamixer.h (handle_source, interface, dir):
* ext/alsa/gstalsamixerelement.c (gst_alsa_mixer_element_details,
gst_alsa_mixer_element_interface_supported,
gst_alsa_mixer_element_finalize, gst_alsa_mixer_element_init,
gst_alsa_mixer_element_set_property,
gst_alsa_mixer_element_get_property,
gst_alsa_mixer_element_change_state):
* ext/alsa/gstalsamixertrack.c (gst_alsa_mixer_track_update):
* gst-libs/gst/interfaces/mixer.c (gst_mixer_volume_changed,
gst_mixer_option_changed):
* gst-libs/gst/interfaces/mixer.h (set_option, get_option,
volume_changed, option_changed, _gst_reserved):
Implement notification for alsamixer. Fixes #152864
David Schleef [Tue, 15 May 2007 03:53:11 +0000 (03:53 +0000)]
gst/videotestsrc/videotestsrc.*: Add support for video/x-raw-bayer.
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c:
* gst/videotestsrc/videotestsrc.h:
Add support for video/x-raw-bayer.
David Schleef [Sun, 13 May 2007 01:06:19 +0000 (01:06 +0000)]
sys/xvimage/xvimagesink.c: Add some sanity checking for the XVImage size returned by X.
Original commit message from CVS:
* sys/xvimage/xvimagesink.c:
Add some sanity checking for the XVImage size returned by X.
Related to #377400.
Wim Taymans [Sat, 12 May 2007 16:18:39 +0000 (16:18 +0000)]
gst-libs/gst/rtp/gstbasertpdepayload.c: Parse and use additional caps fields as described in updated application/x-rt...
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps),
(gst_base_rtp_depayload_set_gst_timestamp):
Parse and use additional caps fields as described in updated
application/x-rtp caps spec.
Wim Taymans [Sat, 12 May 2007 16:16:22 +0000 (16:16 +0000)]
ext/ogg/gstoggdemux.c: If there is a stream in a chain without any data packets, ignore the stream in the total lengt...
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
(gst_ogg_demux_collect_chain_info):
If there is a stream in a chain without any data packets, ignore the
stream in the total length calculations. Might be related to #436820.
Jan Schmidt [Fri, 11 May 2007 17:33:43 +0000 (17:33 +0000)]
gst/typefind/gsttypefindfunctions.c: Consolidate and re-work our mpeg system stream detection to probe more packets a...
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg_sys_is_valid_pack),
(mpeg_sys_is_valid_pes), (mpeg_sys_is_valid_sys),
(mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
(mpeg_video_type_find), (mpeg_video_stream_type_find),
(plugin_init):
Consolidate and re-work our mpeg system stream detection to probe
more packets and produce a higher confidence result. Fixes a
regression caused by lowering the typefind probability last year
- related to bug #397810. Remove the redundant MPEG-1 specific
typefind function, as the new one detects both MPEG-1 & MPEG-2
happily.
Also cleanup the MPEG elementary and MPEG-TS detection functions a
little.
Tested against my media test directory, with some improvements and
no regressions.
Wim Taymans [Thu, 10 May 2007 15:28:13 +0000 (15:28 +0000)]
gst/playback/gstplaybasebin.c: Connect to the new queue "pushing" signal instead of the broken "running" one.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (fill_buffer), (check_queue),
(queue_out_of_data):
Connect to the new queue "pushing" signal instead of the broken
"running" one.
Sébastien Moutte [Wed, 9 May 2007 21:17:40 +0000 (21:17 +0000)]
gst-libs/gst/rtp/gstbasertpaudiopayload.c: Move variable declaration before the first instruction.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
(gst_base_rtp_audio_payload_handle_frame_based_buffer):
Move variable declaration before the first instruction.
* gst/videotestsrc/videotestsrc.c:
Define M_PI if it's not defined yet.
* win32/common/libgstrtp.def:
Add new exported functions.
Michael Smith [Wed, 9 May 2007 11:54:32 +0000 (11:54 +0000)]
ext/theora/theoradec.c: gst_pad_push_event() does not return a GstFlowReturn!
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_handle_type_packet):
gst_pad_push_event() does not return a GstFlowReturn!
Wim Taymans [Wed, 9 May 2007 11:25:34 +0000 (11:25 +0000)]
tests/examples/seek/: Some small cosmetic changes.
Original commit message from CVS:
* tests/examples/seek/scrubby.c: (stop_cb), (main):
* tests/examples/seek/seek.c: (do_seek):
Some small cosmetic changes.
Stefan Kost [Tue, 8 May 2007 19:24:01 +0000 (19:24 +0000)]
gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected, gst_adder_change_state): gst/adder/gstadder.h (bps, o...
Original commit message from CVS:
* gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected,
gst_adder_change_state):
* gst/adder/gstadder.h (bps, offset, collect_event, segment,
segment_pending, segment_position, segment_rate):
Handle playback-rate on adder.
Michael Smith [Mon, 7 May 2007 11:43:31 +0000 (11:43 +0000)]
ext/theora/: Don't push events (newsegment, tags) before initialising the decoder.
Original commit message from CVS:
* ext/theora/gsttheoradec.h:
* ext/theora/theoradec.c: (gst_theora_dec_reset),
(theora_dec_sink_event), (theora_handle_comment_packet),
(theora_handle_type_packet), (theora_dec_change_state):
Don't push events (newsegment, tags) before initialising the
decoder.
This is neccesary for seeking to work correctly in gnonlin.
Stefan Kost [Fri, 4 May 2007 13:10:07 +0000 (13:10 +0000)]
gst/: gst/audiotestsrc/gstaudiotestsrc.c
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst/adder/gstadder.c:
* gst/audiotestsrc/gstaudiotestsrc.c
(gst_audio_test_src_create_white_noise):
* gst/videotestsrc/gstvideotestsrc.c:
* gst/volume/gstvolume.c (VOLUME_UNITY_INT16,
VOLUME_UNITY_INT16_BIT_SHIFT, VOLUME_MAX_DOUBLE,
volume_sink_template, volume_src_template, gst_volume_init,
volume_process_double, volume_process_int16,
volume_process_int16_clamp):
Doc fixes and formatting.
Tim-Philipp Müller [Fri, 4 May 2007 12:41:21 +0000 (12:41 +0000)]
tests/check/: Minimal check for volume's GstController usability; also another test for #422295.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
Minimal check for volume's GstController usability; also another
test for #422295.
Tim-Philipp Müller [Fri, 4 May 2007 09:06:38 +0000 (09:06 +0000)]
gst-libs/gst/cdda/gstcddabasesrc.c: Fix it so that it (a) makes sense and (b) doesn't break everything cdda-related i...
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_add_track):
Fix it so that it (a) makes sense and (b) doesn't break
everything cdda-related including the unit test.
Stefan Kost [Fri, 4 May 2007 08:46:59 +0000 (08:46 +0000)]
gst-libs/gst/cdda/gstcddabasesrc.c: Fix build when disabling asserts.
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_add_track):
Fix build when disabling asserts.
Tim-Philipp Müller [Thu, 3 May 2007 16:29:10 +0000 (16:29 +0000)]
sys/ximage/ximagesink.c: When XShm is not available, we might get row strides that are not rounded up to multiples of...
Original commit message from CVS:
* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new):
When XShm is not available, we might get row strides that are not
rounded up to multiples of four; this is bad, because virtually
every RGB-processing element in GStreamer assumes rowstrides are
rounded up to multiples of four, so let's allocate at least enough
memory to avoid crashes in this case. The image will still be
displayed distorted though if this happens, so that still needs
fixing (maybe by allocating a bigger image with an 'even' width
and then clipping it appropriately when rendering - something for
Xlib aficionados in any case).
Michael Smith [Thu, 3 May 2007 13:16:21 +0000 (13:16 +0000)]
gst/audiorate/gstaudiorate.c: If a buffer doesn't have a timestamp, assume it's contiguous with the previous buffer, ...
Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
If a buffer doesn't have a timestamp, assume it's contiguous with
the previous buffer, and synthesise timestamps appropriately.
Edward Hervey [Thu, 3 May 2007 11:24:00 +0000 (11:24 +0000)]
tests/check/elements/videorate.c: Set buffer timestamp to a valid value in order to test the buffer really does stay ...
Original commit message from CVS:
* tests/check/elements/videorate.c: (GST_START_TEST):
Set buffer timestamp to a valid value in order to test the buffer
really does stay in videorate.
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
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