Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* gst/videorate/gstvideorate.c: (gst_video_rate_reset),
(gst_video_rate_flush_prev), (gst_video_rate_event),
(gst_video_rate_chain):
* gst/videorate/gstvideorate.h:
React (more) to NEWSEGMENT
Small adjustment in timestamp calculation to prevent mismatches
Fixes#435633.
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query_duration):
Correctly resync the iterator if gst_iterator_next() returns
GST_ITERATOR_RESYNC.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
(gst_audio_clock_reset), (gst_audio_clock_get_internal_time):
* gst-libs/gst/audio/gstaudioclock.h:
Add method to inform the clock that the time starts from 0 again. We use
this info to calculate a clock offset so that the time we report in
internal_time is monotonically increasing, as required by the clock base
class. Fixes#521761.
API: GstAudioClock::gst_audio_clock_reset()
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_change_state):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
Reset reported time when we (re)create the ringbuffer.
Original commit message from CVS:
* ext/alsa/gstalsamixertrack.c:
(gst_alsa_mixer_track_update_alsa_capabilities):
Make sure playback volumes aren't accidentally overwritten by
capture volumes if an alsa mixer track has both playback and
capture capabilities: we create two GstMixerTracks in that
case, so make sure we query only the alsa capabilities that
refer to the type of GstMixerTrack we created from the dual
capability alsa element. Should fix issues with Audigy2 sound
cards (#518082).
Original commit message from CVS:
* gst/playback/gstdecodebin2.c:
(gst_decode_group_control_source_pad), (gst_decode_group_expose):
Check for NULL cases and log them, creating ghostpads can, for example,
fail when the pad returns wrong caps.
* gst/playback/gstplaybin2.c: (perform_eos):
When pushing out the EOS event, collect the return value and warn when
something failed.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add support for DVCPRO.
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c: (DEFAULT_PROP_METHOD):
Change default scaling method from nearest-neighbour to bilinear.
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (parser_state_init),
(gst_sub_parse_format_autodetect), (handle_buffer):
* gst/subparse/gstsubparse.h:
* tests/check/elements/subparse.c: (test_tmplayer_style3b):
Limit duration to a maximum of five seconds for tmplayer format where
we can guess the duration only from the timestamp of the next line of
text. We don't want to show a text for eternities just because nothing
else is being said for a while.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_handle_sink_event),
(gst_base_rtp_depayload_push_full),
(gst_base_rtp_depayload_change_state):
Check sequence numbers, mark input buffers with a discont flag for the
subclass when we detected a gap, drop duplicate buffers. We do this
because one can use the element without a jitterbuffer in front and we
don't want to feed the subclasses invalid or reordered data.
Do an error when the subclass did not provide a process function instead
of crashing.
Some other small cleanups.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/playback/gstdecodebin.c: (remove_fakesink):
Lock the fakesink before setting the state to NULL and removing it from
the bin so that a concurrent state change cannot interfere.
Fixes#534331.
Original commit message from CVS:
2008-05-21 Julien Moutte <julien@fluendo.com>
* gst/tcp/gstmultifdsink.c: (setup_dscp_client): Use IPPROTO_IP
instead of SOL_IP, works on more platforms.
* gst/typefind/gsttypefindfunctions.c: (aac_type_find): Fix printf
arguments.
Original commit message from CVS:
* ext/vorbis/vorbisdec.c:
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_show_frame):
Some debug and comment fixes.
* tests/examples/dynamic/addstream.c: (main):
Fix , to ;
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
Fix wrong method name in docs. Fix calculation of strf fields for
broken mulaw/alaw.
* gst-libs/gst/riff/riff-read.c:
Whitespace fix and removing double ';'.
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to
other channel positions when source has SIDE channels and dest doesn't
or the other way around.
Original commit message from CVS:
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add another test that checks if conversion between standard 1 and 2
channel layouts with and without positions set is working.
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions):
Allow non-standard 2 channel layouts.
* tests/check/elements/audioconvert.c: (GST_START_TEST):
Add some tests for converting and remapping non-standard 1 and 2
channel layouts.
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c:
(gst_channel_mix_fill_normalize):
Prevent division by zero if the channel mix matrix contains only
zeroes.
Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst/gdp/gstgdppay.c: (gst_gdp_pay_chain):
Close a buffer memory leak. Fixes bug #534071.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtsptransport.h:
Make the GstRTSPTransport struct members public as there are no
setters/getters and it's supposed to be changed directly.
Fixes bug #533087.
Original commit message from CVS:
* gst/adder/gstadder.c:
Adder also doesn't support audio/x-raw-int with width!=depth so don't
claim this on the pad template caps.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_sync_latency):
We can only use our optimal calibration if we prerolled before the
latency expired.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_channels):
Passthrough the channel positions if the number of output channels is
the same as the number of input channels, the input had a channel
layout and downstream requests no special one. We did this already for
> 2 channels but now it's also done for 1 channel. Fixes bug #533617.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
(gst_gnome_vfs_src_finalize),
(gst_gnome_vfs_src_received_headers_callback),
(gst_gnome_vfs_src_create), (gst_gnome_vfs_src_stop):
* ext/gnomevfs/gstgnomevfssrc.h:
Set the ICY caps on the srcpad from where they get picked up by the base
class now and set on the outgoing buffers.
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
BaseSrc now sets the caps on outgoing buffers automatically.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_resample_slaving),
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
(gst_base_audio_sink_async_play),
(gst_base_audio_sink_change_state):
Change the way in which the ringbuffer is started when dealing with a
slaved clock and latency. We now sync to the clock until we reach
upstream latency before starting the ringbuffer. This has the effect
that we can accurately align the master and slave clocks and let the
rate correction code take care of the initial drift or rounding errors
instead of leaving them uncorrected with the old approach.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_fixate_channels):
Correctly set the default channel positions when converting to 8
channels.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (aac_type_find):
Use data scan helper in aac typefinder and stop scanning
for headers when we've found a type. Also fix potential invalid
memory access when calculating the frame length.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (data_scan_ctx_ensure_data),
(mpeg_sys_is_valid_pack):
Don't modify scan context when we return FALSE in ensure_data, so
it's possible to continue scanning, and we don't end up with a NULL
data pointer and a positive size, which might bite us the next time
we're called. Small constification.
Original commit message from CVS:
* gst/adder/gstadder.c:
Adder doesn't support 24 bit samples so don't claim it supports them
in the pad template caps.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_chain):
Validate the RTP packet before further processing it. It's just too
dangerous to accept random packets and people are not forced to use a
jitterbuffer or session manager to filter out the bad packets.
* gst-libs/gst/rtp/gstrtpbuffer.c:
(gst_rtp_buffer_set_extension_data),
(gst_rtp_buffer_get_payload_subbuffer):
Small cleanups.
When setting extension data in a buffer that is too small, we fail and
we should not set the extension bit.
Change GST_WARNINGS into g_warning because they really are
programming errors.
* tests/check/libs/rtp.c: (GST_START_TEST):
Catch the g_warnings now in the unit tests and that fact that failing to
set extension data left the extension bit untouched.
Original commit message from CVS:
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
Revert previous change which made basetransform handle buffer_alloc
and which breaks things badly in the non-passthrough case since it
returned buffers with a different (ie. sometimes smaller) size than
the size requested.
Original commit message from CVS:
Patch by: Bernard B <b-gnome at largestprime dot net>
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum):
Fix seqnum compare function for bordercase values and fix the docs
again. Fixes#533075.
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
Add a testcase for seqnum compare function.
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_setcaps),
(gst_adder_class_init):
Correctly declare the supported endianness on the pad templates
and check for correct endianness in the set caps function. Adder
only supports native endianness.
Also use gst_element_class_set_details_simple().
Original commit message from CVS:
* sys/xvimage/xvimagesink.c:
Better debug logging in port value handling. Merging separate port
value loops into one.
Original commit message from CVS:
Patch by: Hannes Bistry <hannesb at gmx dot de>
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
* gst/tcp/gsttcpserversink.c:
(gst_tcp_server_sink_handle_server_read),
(gst_tcp_server_sink_handle_wait), (gst_tcp_server_sink_init_send):
Fix regression in clientsrc because we did not add the fd to the poll
set anymore. Fixes#532364.
Do some cleanups here and there.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
* gst/playback/gstplay-marshal.list:
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
Use correct marshallers. GstCaps are a boxed type and no GObject
subclass.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* tests/check/elements/audioresample.c:
(live_switch_alloc_only_48000), (live_switch_get_sink_caps),
(live_switch_push), (GST_START_TEST):
Add unit test for the latest basetransform negotiation changes.
See bug #526768.
Original commit message from CVS:
Patch by: j^ <j at oil21 dot org>
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fishead),
(gst_ogg_pad_parse_skeleton_fisbone):
* ext/ogg/gstoggdemux.h:
Parse presentation time from skeleton streams and use it as offset
for the timestamps. Fixes bug #530068.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
Revert previous patch that attempted to more accurately calculate the
initial offset between master and slave clock. The best thing we can do
in general is take the time of both clocks as the diff since we don't
know when the actual preroll happened.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
Don't do lots of 4-byte peeks, but use the 'new' data scan helper
for this instead; don't check if we've found enough markers after
each and every step, it's enough to do that only if we've actually
found a new marker.
Embed a G_UNLIKELY into the IS_MPEG_HEADER macro.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c:
(DATA_SCAN_CTX_CHUNK_SIZE), (DataScanCtx), (data_scan_ctx_advance),
(data_scan_ctx_ensure_data), (GST_MPEGVID_TYPEFIND_TRY_SYNC),
(mpeg_video_stream_type_find):
Move scan helper thingy to the beginning of the file so we can use
it in other typefind functions. Rename it to something more
generic. Also improve handling of things towards the end of the
typefind data: peek as much as we can if we know the size of the
data, rather than just min_size.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/interfaces/colorbalance.c:
* gst-libs/gst/interfaces/colorbalance.h:
* gst-libs/gst/interfaces/colorbalancechannel.c:
* gst-libs/gst/interfaces/colorbalancechannel.h:
* gst-libs/gst/interfaces/tuner.c:
* gst-libs/gst/interfaces/tunerchannel.c:
* gst-libs/gst/interfaces/tunerchannel.h:
* gst-libs/gst/interfaces/tunernorm.c:
* gst-libs/gst/interfaces/tunernorm.h:
* gst-libs/gst/video/video.c:
* gst-libs/gst/video/video.h:
Document the GstTuner and GstColorBalance interfaces, and some
other random API functions that needed it. 70% symbol coverage, woo.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
Choose to allocate one less segment but require one additional segment
as latency.
* gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_acquire):
No need to increment the number of segments in the source.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_time), (clock_convert_external),
(gst_base_audio_sink_resample_slaving),
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_none_slaving), (gst_base_audio_sink_render),
(gst_base_audio_sink_async_play):
Remove adding latency when returning the internal time while subtracting
it again when we use the value a little later.
When calculating the end timestamp, we are making a rounding error
with the current algorithm. Ensure that we don't accumulate these
rounding errors when aligning samples by not resampling at all if we
don't need to. Fixes#419351.
Make the initial calibration of the clock slaving a little more
predictable and accurate. Also handle the case where we don't do
clock slaving.
Original commit message from CVS:
Based on a patch by:
Björn Benderius <bjoern dot benderius at axis dot com>
* gst/ffmpegcolorspace/avcodec.h:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
(gst_ffmpegcsp_avpicture_fill):
* gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
* gst/ffmpegcolorspace/imgconvert_template.h:
Add conversions from/to NV12 and NV21 and conversions between those
two formats. Fixes bug #532166.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
Abort the h264 typefinding as soon as _peek() doesn't return anything,
which happens for example with files smaller than 128kb.
Original commit message from CVS:
Patch by: Wouter Cloetens <zombie at e2big dot org>
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_create), (md5_digest_to_hex_string),
(auth_digest_compute_hex_urp), (auth_digest_compute_response),
(add_auth_header), (gst_rtsp_connection_free),
(gst_rtsp_connection_set_auth), (str_case_hash), (str_case_equal),
(gst_rtsp_connection_set_auth_param),
(gst_rtsp_connection_clear_auth_params):
* gst-libs/gst/rtsp/gstrtspconnection.h:
Add Digest authorization support for RTSP connections. See #532065.
* gst-libs/gst/rtsp/md5.c:
* gst-libs/gst/rtsp/md5.h:
Yeap, another md5 implementation until we can depend on a glib that has
support for it.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
Let audioresample use the buffer allocation of basetransform instead
of it's own stuff.
* tests/check/elements/audioresample.c: (alloc_only_48000),
(GST_START_TEST), (audioresample_suite):
Add unit test for the recent basetransform bugfix, where upstream
changes caps to something that can't be passed through anymore.
Original commit message from CVS:
* win32/common/config.h.in:
Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather
use the real thing than having "???" unconditionally.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_query):
Report the latency with the new seglatency parameter.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_debug_spec_buff), (gst_ring_buffer_parse_caps),
(gst_ring_buffer_acquire):
* gst-libs/gst/audio/gstringbuffer.h:
Add new field to the ringbufferspec to specify the expected latency
between the underlying device read/write pointer, this is needed
when writing sinks that sit a little closer to the hardware.
Add some more docs for other fields.
Original commit message from CVS:
* gst/volume/gstvolume.c: (volume_transform_ip):
Return NOT_NEGOTIATED if we didn't set a process function yet for some
reason instead of crashing later. Might fix bug #509125.
Original commit message from CVS:
Based on a patch by: Tim-Philipp Müller <tim.muller at collabora co uk>
* gst/audioconvert/audioconvert.c: (audio_convert_prepare_context):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps),
(structure_has_fixed_channel_positions),
(gst_audio_convert_transform_caps):
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_matrix):
Add support for more than 8 channels and NONE channel layouts. For
more than 8 channels no channel conversion is supported yet, only
format conversions are supported. Fixes bug #398033.
* tests/check/elements/audioconvert.c: (verify_convert),
(GST_START_TEST), (audioconvert_suite):
Add some unit tests by Tim for checking the NONE channel layouts
and more than 8 channels and add some more unit tests for channel
conversions.
Original commit message from CVS:
* gst/playback/gstdecodebin2.c: (connect_pad):
When autoplugging fails, set the element back to NULL before
unreffing it.
Original commit message from CVS:
* gst/subparse/samiparse.c: (handle_start_sync),
(end_sami_element), (characters_sami):
Remove trailing, leading and double whitespaces.
Correctly timestamp buffers and output the last buffer too.
* tests/check/elements/subparse.c: (GST_START_TEST),
(subparse_suite):
Add a simple unit test for SAMI parsing.
Original commit message from CVS:
Patch by: Young-Ho Cha <ganadist at chollian dot net>
* gst/subparse/samiparse.c: (handle_start_sync),
(start_sami_element), (end_sami_element), (characters_sami),
(sami_context_reset):
Only output characters inside the "sync" elements. There could be
other elements like "style" that have some content but should
not be printed. Fixes bug #467911.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (set_audio_mute),
(set_active_source):
* gst/playback/gstplaybasebin.h:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
(playbin_set_audio_mute):
Allow setting -1 as current-audio to mute the current audio stream,
similar to what is done for subtitles. Fixes bug #342294.
Original commit message from CVS:
* tests/check/elements/subparse.c: (do_test),
(test_tmplayer_style3b), (subparse_suite):
Add unit test for the tmplayer variant from bug #530962.
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (handle_buffer),
(gst_sub_parse_sink_event):
* gst/subparse/tmplayerparse.c: (tmplayer_process_buffer),
(tmplayer_parse_line):
Fix parsing of tmplayer subtitle variant where every single line contains
text and there isn't an empty line after each line to determine the
duration (#530962). Improve EOS handling for tmplayer subtitles a bit by
making sure that we push out the last line of text without a duration if
there's still text left in the buffer at the end.
Original commit message from CVS:
* gst/subparse/gstsubparse.c: (feed_textbuf):
Fix detection of discontinuities based on the buffer offset (doesn't work
so well if no buffer offset is set) and also check for the DISCONT buffer
flag. This keeps the parser state from being reset after each buffer in
the unit test.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (mpeg_video_stream_type_find):
Further fine-tuning: don't absolutely require sequence or GOP headers
(as introduced in the previous commit), but adjust the typefind
probabilities returned accordingly if we don't see them. Also make sure
picture header and first slice are somewhat close to each other (which
is not perfect but still better than requiring a fixed offset or having
no limit at all).
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
(gst_basertppayload_sink_setcaps),
(gst_basertppayload_sink_getcaps):
Rename the setcaps/getcaps function internally to make it clear that
they are called for the sink pad.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init),
(gst_base_rtp_depayload_handle_sink_event), (create_segment_event),
(gst_base_rtp_depayload_packet_lost),
(gst_base_rtp_depayload_set_gst_timestamp):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Catch packet-lost events from the jitterbuffer and convert them into a
vmethod call (lost-packet) so that depayloaders can do something smart.
Also add a default packet-lost function that sends out a segment update
to the decoders.
Original commit message from CVS:
* gst/playback/test4.c:
* gst/playback/test5.c:
* gst/playback/test6.c:
* gst/playback/test7.c:
Also include config.h when relying on defines from it. Fixes the
build. Its been a please to serve :)
Original commit message from CVS:
* gst/videotestsrc/videotestsrc.c (paint_setup_NV12),
(paint_setup_NV21), (paint_hline_NV12_NV21):
Add support for NV12 and NV21 in videotestsrc
Original commit message from CVS:
* gst/videoscale/gstvideoscale.c:
* gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
* gst/videoscale/vs_image.c: (vs_image_scale_nearest_RGBA),
(vs_image_scale_linear_RGBA), (vs_image_scale_nearest_RGB),
(vs_image_scale_linear_RGB), (vs_image_scale_nearest_YUYV),
(vs_image_scale_linear_YUYV), (vs_image_scale_nearest_UYVY),
(vs_image_scale_linear_UYVY), (vs_image_scale_nearest_Y),
(vs_image_scale_linear_Y), (vs_image_scale_nearest_RGB565),
(vs_image_scale_linear_RGB565), (vs_image_scale_nearest_RGB555),
(vs_image_scale_linear_RGB555):
Support 1x1 images as input and output as for example the BBC HQ new
streams have 1x1 GIFs in the playlists for some reason.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (free_pad_probe_for_element),
(try_to_link_1):
If we can't activate one of the decoders we plugged in (such as,
say, musepackdec) for some reason (it might not support push mode,
for example), remove any pad probes that close_pad_link() might
have set up. This makes sure we later don't try to remove a probe
for a pad that doesn't exist any longer, and avoids nast warnings
and probably other things too.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c:
(mpeg_video_stream_ctx_ensure_data), (mpeg_video_stream_type_find),
(plugin_init):
Rework mpeg video stream typefinding a bit more: make sure sequence,
GOP, picture and slice headers appear in the order they should and
that we've in fact at least had one of each; fix picture header
detection; decouple picture and slice header check - don't assume
they're at a fixed offset, there may be extra data in between. Also,
announce varying degrees of probability depending on what we found
exactly (multiple pictures, at least one picture, just sequence and
GOP headers). Finally, in _ensure_data(), take into account that we
might be typefinding smaller amounts of data, such as the first
buffer of a stream, so fall back to the minimum size needed as long
as that's available, instead of erroring out if there's less than
2kB of data. Fixes#526173. Conveniently also doesn't recognise the
fuzzed file from #399342 as valid.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (MpegVideoStreamCtx),
(mpeg_video_stream_ctx_advance), (mpeg_video_stream_ctx_ensure_data),
(mpeg_video_stream_type_find):
Refactor a bit: use context structure to track parsing offset and size of
available data and make the code a bit clearer. Fixes bad memory access
in #356937.
Original commit message from CVS:
* gst/playback/test4.c:
* gst/playback/test5.c:
* gst/playback/test6.c:
* gst/tcp/gstmultifdsink.c:
Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro
is defined.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.h:
Clarify some docs.
* gst-libs/gst/audio/gstbaseaudiosrc.c: (slave_method_get_type),
(gst_base_audio_src_class_init), (gst_base_audio_src_init),
(gst_base_audio_src_set_slave_method),
(gst_base_audio_src_get_slave_method),
(gst_base_audio_src_set_property),
(gst_base_audio_src_get_property), (gst_base_audio_src_create):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
Add property and methods for selecting the clock slave method in the
source, like in the sink.
We only implement "none" and "re-timestamp" for now.
API: gst_base_audio_src_set_slave_method()
API: gst_base_audio_src_get_slave_method()
Original commit message from CVS:
* ext/alsa/gstalsadeviceprobe.c:
(gst_alsa_get_device_list): Don't return before freeing up
the allocated structures.
Original commit message from CVS:
* ext/ogg/gstoggmux.c:
Update Ogg/Dirac muxing. Removes the weird "KW-DIRAC" bos
packet. Should conform to what we currently think is the
final Ogg/Dirac muxing spec.
