Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c:
(gst_rtp_bin_propagate_property_to_jitterbuffer),
(gst_rtp_bin_change_state), (new_payload_found),
(new_ssrc_pad_found):
Break out of callbacks when we are shutting down.
Make sure no state changes can happen when we reconfigure.
Original commit message from CVS:
* configure.ac:
Require CVS core and base for new audio clock reset method.
* ext/alsaspdif/alsaspdifsink.c: (alsaspdifsink_change_state):
Reset the audio clock. See #521761.
Original commit message from CVS:
* ext/jack/gstjackaudiosink.c:
(gst_jack_audio_sink_allocate_channels):
Include the element name in the port name to avoid duplicate port names.
Original commit message from CVS:
* gst/interleave/deinterleave.c:
Add another example launch line.
* gst/interleave/interleave.c: (interleave_24),
(gst_interleave_finalize), (gst_interleave_base_init),
(gst_interleave_class_init), (gst_interleave_init),
(gst_interleave_request_new_pad), (gst_interleave_release_pad),
(gst_interleave_change_state), (__remove_channels),
(__set_channels), (gst_interleave_sink_getcaps),
(gst_interleave_set_process_function),
(gst_interleave_sink_setcaps), (gst_interleave_sink_event),
(gst_interleave_src_query_duration), (gst_interleave_src_query),
(forward_event_func), (forward_event), (gst_interleave_src_event),
(gst_interleave_collected):
* gst/interleave/interleave.h:
Major rewrite of interleave using GstCollectpads. This new version
also supports almost all raw audio formats and has better caps
negotiation. Fixes bug #506594.
Also update docs and add some more examples.
* tests/check/elements/interleave.c: (interleave_chain_func),
(GST_START_TEST), (src_handoff_float32), (sink_handoff_float32),
(interleave_suite):
Add some more extensive unit tests for interleave.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
When checking the seqnum, reset the jitterbuffer if the gap is too big,
we need to do this so that we can better handle a restarted source.
Fix some comments.
* gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew),
(rtp_jitter_buffer_insert):
Tweak the skew resync diff.
Use our working seqnum compare function in -base.
Rework the jitterbuffer insert code to make it clearer and more
performant by only retrieving the seqnum of the input buffer once and by
adding some G_LIKELY compiler hints.
Improve debugging for duplicate packets.
* gst/rtpmanager/rtpsource.c: (rtp_source_process_rtp):
Fix a comment, we don't do skew correction here..
Original commit message from CVS:
Patch by: Håvard Graff <havard dot graff at tandberg dot com>
* gst/rtpmanager/gstrtpbin.c:
(gst_rtp_bin_propagate_property_to_jitterbuffer),
(gst_rtp_bin_set_property):
Propagate the do-lost and latency properties to the jitterbuffers when
they are changed on rtpbin.
Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_base_init),
(gst_deinterleave_class_init), (gst_deinterleave_init),
(gst_deinterleave_add_new_pads), (gst_deinterleave_sink_getcaps):
* gst/interleave/deinterleave.h:
Don't set a getcaps() function on the src pads as it's not required
and the default getcaps() function returns the correct results for
our src pads.
Complete documentation and add myself to the authors of the element.
Original commit message from CVS:
* tests/icles/.cvsignore:
* tests/icles/Makefile.am:
* tests/icles/test-oss4.c: (opt_show_mixer_messages), (WAIT_TIME),
(show_mixer_messages), (probe_mixer_tracks), (probe_pad),
(probe_details), (probe_element), (main):
Small oss4 test that probes for available devices and retrieves
their caps and mixer tracks and all that. Also allows testing of
mixer change messages on the bus.
Original commit message from CVS:
* sys/oss4/oss4-mixer.c: (gst_oss4_mixer_open):
* sys/oss4/oss4-property-probe.c:
(gst_oss4_property_probe_find_device_name),
(gst_oss4_property_probe_find_device_name_nofd):
* sys/oss4/oss4-property-probe.h:
* sys/oss4/oss4-sink.c: (gst_oss4_sink_get_property):
* sys/oss4/oss4-source.c: (gst_oss4_source_get_property):
Make device-name probing in NULL state work better (e.g. for the
gnome-control-center sound capplet).
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/mpeg4videoparse/mpeg4videoparse.c: (gst_mpeg4vparse_push),
(gst_mpeg4vparse_drain), (gst_mpeg4vparse_chain),
(gst_mpeg4vparse_change_state):
Move some code around to integrate the startcode searching with the
other bits of parsing, avoid a whole bunch of peeks.
Get rid of invalid data that should not happen according to the specs.
Fixes#533559.
Original commit message from CVS:
Patch by: Bastien Nocera <hadess at hadess dot net>
* ext/mythtv/gstmythtvsrc.c: (gst_mythtv_src_class_init),
(gst_mythtv_src_init), (gst_mythtv_src_clear),
(do_read_request_response), (gst_mythtv_src_create),
(gst_mythtv_src_start):
Correctly set duration to get a more correct seek bar in totem.
Disable query and event functions as they don't work and do some
smaller cleanup.
Fixes bug #533736.
Original commit message from CVS:
* tests/check/elements/deinterleave.c: (GST_START_TEST):
Set keep-positions property to TRUE for the 8 channel test to ensure
that the original channel position is set on the output.
Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_class_init),
(gst_deinterleave_init), (gst_deinterleave_add_new_pads),
(gst_deinterleave_set_pads_caps), (gst_deinterleave_set_property),
(gst_deinterleave_get_property):
* gst/interleave/deinterleave.h:
Add a property to select whether channel positions should be kept on
the mono output buffers or should be dropped.
Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_finalize),
(gst_deinterleave_init), (gst_deinterleave_sink_event),
(gst_deinterleave_process), (gst_deinterleave_sink_activate_push):
* gst/interleave/deinterleave.h:
Queue events until src pads were added and they can be sent. Otherwise
downstream will never get the first newsegment event.
Original commit message from CVS:
* gst/interleave/deinterleave.c: (gst_deinterleave_sink_setcaps),
(gst_deinterleave_getcaps):
Always set the channel positions when gst_audio_get_channel_positions()
returns something, even if they're not set in the caps. This makes
sure that the output channels can be interleaved again correctly
in the mono/stereo cases too.
Don't ask for the peercaps of the current pad in getcaps() as this
might call getcaps() again and deadlock.
Original commit message from CVS:
* ext/timidity/gstwildmidi.c: (wildmidi_open_config):
Check some more common locations for a valid configuration file.
Fixes bug #533435. Packagers should still #define WILDMIDI_CFG
to the distributions default location.
Original commit message from CVS:
* gst/interleave/Makefile.am:
* gst/interleave/deinterleave.c: (deinterleave_24),
(gst_deinterleave_finalize), (gst_deinterleave_base_init),
(gst_deinterleave_class_init), (gst_deinterleave_init),
(gst_deinterleave_add_new_pads), (gst_deinterleave_set_pads_caps),
(gst_deinterleave_set_process_function),
(gst_deinterleave_sink_setcaps), (__remove_channels),
(__set_channels), (gst_deinterleave_getcaps),
(gst_deinterleave_process), (gst_deinterleave_chain),
(gst_deinterleave_sink_activate_push):
* gst/interleave/deinterleave.h:
Add support for all raw audio formats and provide better negotiation
if the caps are changing.
Don't allow changes of the channel positions and set the position of
the corresponding channel on the src pad caps.
General cleanup and smaller bugfixes.
* tests/check/elements/deinterleave.c: (float_buffer_check_probe):
Check the channel positions on the output buffer caps.
Original commit message from CVS:
* docs/Makefile.am:
Don't attempt to build plugin docs when they're disabled.
* gst/bayer/Makefile.am:
Add libgstvideo to the link.
* gst/rtpmanager/Makefile.am:
Fix link order, and move LIBS things to _LIBS
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins.types:
Remove bogus attempt to pull 'metadata' plugin's base
class into the docs.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain):
Simply drop bad RTP packets with a warning instead of just posting an
error and stopping. This is a perfectly recoverable event and we don't
force people to use an rtpbin to filter out bad packets first.
Original commit message from CVS:
* gst/mpeg4videoparse/mpeg4videoparse.c: (gst_mpeg4vparse_init):
Set fixed caps on the srcpad after we created the pad...
Original commit message from CVS:
* tests/check/Makefile.am:
Remove deinterleave test from VALGRIND_TO_FIX again now that
there are suppressions in gst.supp which make this work for me.
Original commit message from CVS:
* tests/check/Makefile.am:
Add deinterleave unit test to VALGRIND_TO_FIX, since it causes
weird invalid free errors in valgrind/libc after _exit for some
reason.
* tests/check/elements/deinterleave.c: (pads_created),
(set_channel_positions), (src_handoff_float32_8ch),
(float_buffer_check_probe),
(pad_added_setup_data_check_float32_8ch_cb),
(make_fake_src_8chans_float32), (GST_START_TEST),
(deinterleave_suite):
Add some more deinterleave unit test bits I had locally.
Original commit message from CVS:
* gst/interleave/Makefile.am:
* gst/interleave/deinterleave.h:
* gst/interleave/interleave.h:
* gst/interleave/plugin.h:
Split definitions into separate header files for better documentation
generation.
* gst/interleave/deinterleave.c: (gst_deinterleave_base_init),
(gst_deinterleave_class_init), (gst_deinterleave_sink_setcaps),
(gst_deinterleave_process):
Don't use alloca, allow caps changes as long as the number of channels
does not change, don't use g_warning, return NOT_NEGOTIATED as early
as possible and some other cleanup.
* gst/interleave/interleave.c: (gst_interleave_base_init),
(gst_interleave_class_init):
Do some random cleanup.
* tests/check/Makefile.am:
* tests/check/elements/deinterleave.c: (GST_START_TEST),
(deinterleave_chain_func), (deinterleave_pad_added),
(deinterleave_suite):
Add unit tests for the deinterleave element.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/mpeg4videoparse/mpeg4videoparse.c:
(gst_mpeg4vparse_set_new_caps), (gst_mpeg4vparse_align),
(get_bits), (next_start_code), (gst_mpeg4vparse_handle_vos),
(gst_mpeg4vparse_push), (gst_mpeg4vparse_drain),
(gst_mpeg4vparse_chain), (gst_mpeg4vparse_sink_setcaps),
(gst_mpeg4vparse_sink_event), (gst_mpeg4vparse_src_query),
(gst_mpeg4vparse_set_property), (gst_mpeg4vparse_get_property),
(gst_mpeg4vparse_class_init), (gst_mpeg4vparse_init):
* gst/mpeg4videoparse/mpeg4videoparse.h:
Parse the config data (either outbound or in the stream) to set
width/height, apect ration, framerate in the caps if applicable.
Mark frames as GST_BUFFER_FLAG_DELTA_UNIT when they are not
intra frames
Set the timestamps of outgoing buffers to the buffer in
which the VOP header was found.
Drop incoming data untill configuration is found (by default,
configurable using a property).
Report a 1 frame latency. Fixes#532723.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_loop):
Avoid waiting for a negative (huge) duration when the last packet has a
lower timestamp than the current packet.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_query_send_rtcp_src):
Make sure to unref the rtpsession returned by gst_pad_get_parent() to
prevent a memory leak.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_loop):
Initialise with GST_CLOCK_TIME_NONE to avoid compiler warning.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/inspect/plugin-gstinterlace.xml:
* gst/deinterlace/gstdeinterlace.c:
* gst/deinterlace/gstdeinterlace.h:
Random doc of the day: the deinterlace element.
Original commit message from CVS:
* gst/mpegtsparse/mpegtspacketizer.c:
* gst/mpegtsparse/mpegtsparse.c:
Make sure all schedule EIT and non-actual transport stream
EITs are parsed. Also add present-following flag and
actual-transport-stream flag to eit bus message.
Original commit message from CVS:
* gst/rtpmanager/rtpsource.c: (rtp_source_finalize):
Make sure to unref the caps used by RTPSource to prevent a memory leak.
Original commit message from CVS:
Based on patch by: Clive Wright <clive_wright ntlworld com>
* sys/oss4/oss4-mixer-slider.c: (gst_oss4_mixer_slider_unpack_volume):
Apparently mono sliders have the mono value repeated in the upper bits,
so mask those out when reading them. Probably makes the mixer applet
work properly in some more cases.
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester dot ca>
* gst/rtpmanager/rtpsession.c: (source_clock_rate),
(rtp_session_process_bye), (rtp_session_send_bye_locked):
Unlock the session lock when calling one of our callbacks.
Fixes#532011.
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/app/gstappsink.c: (gst_app_sink_class_init),
(gst_app_sink_init), (gst_app_sink_set_property),
(gst_app_sink_get_property), (gst_app_sink_unlock_start),
(gst_app_sink_unlock_stop), (gst_app_sink_flush_unlocked),
(gst_app_sink_start), (gst_app_sink_stop), (gst_app_sink_event),
(gst_app_sink_preroll), (gst_app_sink_render),
(gst_app_sink_set_caps), (gst_app_sink_set_drop),
(gst_app_sink_get_drop):
* gst-libs/gst/app/gstappsink.h:
Start some docs.
Add property to drop buffers when the queue is filled
Fix unlocking and flushing when the queues are filled.
Original commit message from CVS:
Patch by: Jens Granseuer <jensgr at gmx dot net>
* gst/subenc/gstsrtenc.c: (gst_srt_enc_timestamp_to_string):
Declare variables at the beginning of blocks. Fixes compilation with
gcc 2.x and other compilers. Fixes bug #530611.
Original commit message from CVS:
* gst/mpegtsparse/mpegtspacketizer.c:
* gst/mpegtsparse/mpegtspacketizer.h:
* gst/mpegtsparse/mpegtsparse.c:
Detect SI pids (NIT, SDT, EIT etc.) based on table id and not
by pid number. This allows for example the EPG data from UK's
freesat to be picked up.
Original commit message from CVS:
* ext/mpeg2enc/gstmpeg2enc.cc:
* ext/soundtouch/gstbpmdetect.cc:
Cast NULL sentinels to void * as NULL is defined as an integer
constant in most environments when using C++ and it's size might
be different from a pointer.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
(gst_app_sink_init), (gst_app_sink_set_property),
(gst_app_sink_get_property), (gst_app_sink_event),
(gst_app_sink_preroll), (gst_app_sink_render),
(gst_app_sink_set_emit_signals), (gst_app_sink_get_emit_signals),
(gst_app_sink_set_max_buffers), (gst_app_sink_get_max_buffers),
(gst_app_sink_pull_buffer):
* gst-libs/gst/app/gstappsink.h:
Add more docs.
Add signals for when preroll and render buffers are available.
Add property to control signal emission.
Add property to control the max queue size.
Original commit message from CVS:
* gst-libs/gst/dshow/Makefile.am:
Use CXXFLAGS rather than CFLAGS; these are C++ files.
Define required constants appropriately.
* sys/dshowdecwrapper/Makefile.am:
Add required include dir, libraries.
Define required constants appropriately.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (create_stream), (gst_rtp_bin_init),
(gst_rtp_bin_set_property), (gst_rtp_bin_get_property):
* gst/rtpmanager/gstrtpbin.h:
Expose new jitterbuffer property in rtpbin too.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_init),
(gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_set_property),
(gst_rtp_jitter_buffer_get_property):
Disable sending out rtp packet lost events by default and make a
property to enabe it. We will likely enable it by default when the base
depayloaders have a default handler for them so that we don't send these
events all through the pipeline for now.
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_flush_stop),
(gst_rtp_jitter_buffer_src_event), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_loop):
Remove private version of a function that is in -base now.
Add src event handler.
Rework the jitterbuffer pushing loop so that it can quickly react to
lost packets and instruct the depayloader of them. This can then be used
to implement error concealment data.
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c:
(gst_rtp_session_query_send_rtcp_src), (create_recv_rtcp_sink),
(create_send_rtcp_src):
Set up some internal links functions for the RTCP and sync pads because
the defaults are really not correct.
Implement a query handler for the RTCP src pad, mostly to correctly
report about the latency.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
(gst_rtp_bin_sync_chain):
* gst/rtpmanager/rtpsession.c: (update_arrival_stats),
(rtp_session_process_sr), (rtp_session_on_timeout):
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(calculate_jitter):
* gst/rtpmanager/rtpsource.h:
* gst/rtpmanager/rtpstats.h:
Also keep track of the first buffer timestamp together with the first
RTP timestamp as they both are needed to construct the timing of
outgoing packets in the jitterbuffer and are therefore also needed to
manage lip-sync. This fixes lip-sync if the first RTP packets arrive
with a wildly different gap.
Original commit message from CVS:
* configure.ac:
* ext/musepack/gstmusepackdec.c: (gst_musepackdec_base_init),
(gst_musepackdec_init), (gst_musepackdec_dispose),
(gst_musepackdec_handle_seek_event), (gst_musepack_stream_init),
(gst_musepackdec_loop), (plugin_init):
* ext/musepack/gstmusepackdec.h:
* ext/musepack/gstmusepackreader.c:
* ext/musepack/gstmusepackreader.h:
Add support for the new libmpcdec API which magically gets us support
for SV8 files. Also do some random cleanup. Fixes bug #526905.
Original commit message from CVS:
* tests/check/Makefile.am:
Don't inlcude dc1394src in the generic/states test as it requires
special hardware. Fixes bug #528011.
Original commit message from CVS:
* tests/check/elements/ofa.c: (bus_handler), (GST_START_TEST):
Only check if the generated fingerprints are valid Base64. The
fingerprints are different when running on different architectures
which is a) no problem because the fingerprints are tolerant enough
and b) is caused by libofa. Fixes bug #528266.
Original commit message from CVS:
* ext/timidity/Makefile.am:
Dist all source files, no matter if only timidity or wildmidi or
nothing is found by configure. Fixes bug #528000.
Original commit message from CVS:
* ext/dirac/gstdiracenc.cc:
Fix compilation by casting string constants.
* sys/Makefile.am:
Fix WININET_DIR variable reference.
Original commit message from CVS:
* gst/flv/gstflvparse.c: (gst_flv_parse_metadata_item),
(gst_flv_parse_tag_script):
Handle NULL returns from FLV_GET_STRING() more gracefully. Fixes
crash caused by a strlen on a NULL string (#527622).
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås <ole.andre.ravnas at tandberg com>
* sys/dshowsrcwrapper/gstdshowvideosrc.c: (PROP_DEVICE_NAME),
(gst_dshowvideosrc_class_init), (gst_dshowvideosrc_init),
(gst_dshowvideosrc_dispose), (gst_dshowvideosrc_stop),
(gst_dshowvideosrc_unlock), (gst_dshowvideosrc_unlock_stop),
(gst_dshowvideosrc_create), (gst_dshowvideosrc_push_buffer):
* sys/dshowsrcwrapper/gstdshowvideosrc.h:
Don't increase latency by queuing buffers in an async queue when
the streaming thread can't keep up or isn't scheduled often
enough for some other reason, but just drop the previous buffer
in that case. Also implement GstBaseSrc::unlock for faster
unlocking when shutting down. (#520892).
Original commit message from CVS:
* tests/icles/metadata_editor.c: (ENC_UNKNOWN), (last_pixbuf),
(draw_pixbuf), (change_tag_list), (update_draw_pixbuf),
(ui_drawing_size_allocate_cb), (on_drawingMain_expose_event),
(on_buttonSaveFile_clicked), (ui_create), (me_gst_bus_callback_view),
(me_gst_setup_view_pipeline), (process_file):
* tests/icles/metadata_editor.glade:
Remove GstXOverlay stuff and use gdkpixbufsink plus some rather crude
drawing/scaling logic to make this compile and work on all platforms.
Fixes#518227.
Original commit message from CVS:
* configure.ac:
Bump core/base requirements to released versions to avoid
confusion.
* gst/deinterlace/gstdeinterlace.c: (deinterlace_debug),
(GST_CAT_DEFAULT), (gst_deinterlace_base_init),
(gst_deinterlace_set_caps), (plugin_init):
Add debug category, use _set_element_details_simple and
remove special code path for Y42B to calculate offsets and
strides; libgstvideo knows how to handle this format now.
Original commit message from CVS:
* gst/cdxaparse/Makefile.am:
* gst/cdxaparse/gstcdxaparse.c:
* gst/cdxaparse/gstcdxastrip.c:
* gst/cdxaparse/gstcdxastrip.h:
* gst/cdxaparse/gstvcdparse.c:
* gst/cdxaparse/gstvcdparse.h:
Port VCD parser (formerly cdxastrip) from 0.8 to 0.10. Doesn't do
anything the 0.8 version didn't do though.
Original commit message from CVS:
Patch by: Ole André Vadla Ravnås <ole.andre.ravnas at tandberg com>
* configure.ac:
* sys/Makefile.am:
* sys/wininet/Makefile.am:
* sys/wininet/gstwininetsrc.c:
* sys/wininet/gstwininetsrc.h:
Add wininetsrc for basic http/ftp support on windows (#520897).
Original commit message from CVS:
* tests/check/elements/souphttpsrc.c: (got_buffer),
(souphttpsrc_suite):
Increase the timeout for the internet tests to 250 seconds
and check for NULL caps instead of just crashing.
The real fix would be to implement an shoutcast server for the unit test
instead of relying on a working internet connection.
Fixes bug #521749.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (free_session):
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_finalize):
Avoid leaking pads in the RTP manager.
Original commit message from CVS:
* ext/jack/gstjackaudiosink.c: (gst_jack_audio_sink_class_init):
Work around missing bits of thread-safety on older GLibs some
more to avoid assertions when starting up multiple playbin
objects concurrently (see #512382).
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* ext/soup/gstsouphttpsrc.c: (gst_soup_http_src_response_cb):
Only ignore actual redirects not all responses when in state
GST_SOUP_HTTP_SRC_SESSION_IO_STATUS_RUNNING. Fixes bug #526337.
Original commit message from CVS:
* tests/check/elements/ofa.c: (GST_START_TEST):
Also check that we have processed at least 135 seconds of audio
until we stop and calculated a fingerprint.
Original commit message from CVS:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/ofa.c: (bus_handler), (GST_START_TEST),
(ofa_suite), (main):
Add simple unit tests for the OFA plugin.
Original commit message from CVS:
* gst/mpegtsparse/mpegtspacketizer.c:
Cable delivery subsystem descriptors' frequency's bcd
is measured in 100Hz units so adjust multiplier accordingly.
Original commit message from CVS:
Patch by: kapil <kapil at fluendo dot com>
* ext/gsm/gstgsmdec.c: (gst_gsmdec_sink_setcaps),
(gst_gsmdec_chain):
* ext/gsm/gstgsmdec.h:
Increase the allowed samplerates for the ms-gsm format.
Fixes#481354.
