Original commit message from CVS:
* gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_create),
(gst_tcpserversrc_start):
Don't block in accept while doing the state change, move
to poll and make cancellable.
Original commit message from CVS:
2005-10-09 Philippe Khalaf <burger@speedy.org>
* gst-libs/gst/rtp/rtpbasedepayload.c:
Set timestamp and add queue delay to timestamp
* gst-libs/gst/rtp/rtpbuffer.h:
Set correct payload type for h263
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (wavpack_type_find),
(plugin_init):
Add wavpack and spc typefind functions from 0.8 branch.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (tar_type_find),
(ar_type_find), (msdos_type_find), (plugin_init):
Add typefind functions for tar archives, ar archives,
RAR archives, and msdos-executables (dlls, exe, etc.).
Some of those would be wrongly identified as mpeg
streams of some sort before (#315550).
Original commit message from CVS:
* gst/adder/gstadder.c: (gst_adder_query), (gst_adder_class_init),
(gst_adder_init), (gst_adder_request_new_pad),
(gst_adder_change_state):
Add query function to source pad, so adder reports the correct
time/sample position when queried (#315457); fix state change
function; use GST_DEBUG_FUNCPTR() for pad functions.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (utf8_type_find):
Fix leaks in typefind registration
Clean up the gratuitous commenting and whitespacing a little
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_stop):
Only actually wait for the thread to be stopped if it's
running.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_render):
If we receive EOS we can start playback of what we had.
Original commit message from CVS:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init),
(gst_multifdsink_finalize), (multifdsink_hash_remove),
(gst_multifdsink_stop):
Fix crasher when going to NULL multiple times.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_event),
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_read):
patch from Edgard Lima <edgard.lima@indt.org.br>
Fixed gstbaseaudiosrc adding ring buffer sync to it.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_loop):
Report the FLOW_RETURN as string in the error message.
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_clear_all):
Don't assert when clearing an unnegotiated buffer.
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (group_destroy),
(gen_preroll_element), (remove_groups), (setup_source):
* gst/playback/gstplaybin.c: (remove_sinks), (add_sink),
(setup_sinks), (gst_play_bin_send_event),
(gst_play_bin_change_state):
Set state to NULL before removing from bin. Fix refcounting.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gst_play_bin_send_event):
Correct refcounting in send_event() function. Previously was wrong
if the first sink was unable to handle the event.
Original commit message from CVS:
2005-10-03 Andy Wingo <wingo@pobox.com>
* gst/playback/gstdecodebin.c (try_to_link_1)
(remove_element_chain): set element to NULL before removing it.
Original commit message from CVS:
* ext/gnomevfs/gstgnomevfssrc.c (gst_gnomevfssrc_uri_get_protocols):
protect gst_gnomevfs_get_supported_uris by a mutex, to make it
MT safe.
Original commit message from CVS:
2005-10-02 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstringbuffer.c (gst_ring_buffer_clear)
(gst_ring_buffer_prepare_read):
* gst-libs/gst/audio/gstaudiosink.c (audioringbuffer_thread_func):
Demote to LOG.
Original commit message from CVS:
2005-09-28 Andy Wingo <wingo@pobox.com>
* gst/videotestsrc/gstvideotestsrc.c: Implement live source mode
and unlocking.
Original commit message from CVS:
2005-09-28 Andy Wingo <wingo@pobox.com>
* gst/tcp/gsttcpclientsink.c (gst_tcpclientsink_base_init):
Actually add the pad template.
(gst_tcpclientsink_get_type): We're a base sink. Woot, works.
* gst/tcp/gsttcpserversrc.c: Go ahead and fix up serversrc while
I'm at it...
Original commit message from CVS:
2005-09-28 Andy Wingo <wingo@pobox.com>
* gst/tcp/gsttcpclientsrc.c: Make interruptable -- code stolen
from fdsrc. Get caps in create() instead of start() so it can be
interrupted. Interruption somewhat untested.
* gst/tcp/gsttcp.c (gst_tcp_read_buffer, gst_tcp_socket_read):
Proper EOS handling.
Original commit message from CVS:
2005-09-27 Andy Wingo <wingo@pobox.com>
* gst/tcp/gsttcpserversrc.c:
* gst/tcp/gsttcpclientsrc.c: Updated for new gsttcp API.
* gst/tcp/gsttcp.h:
* gst/tcp/gsttcp.c (gst_tcp_read_buffer): New function, factored
out of tcpclientsrc.c. Cancellable.
(gst_tcp_socket_read): Made private, cancellable, with better
diagnostics. Also the FIONREAD ioctl takes a int*, not a size_t*.
(gst_tcp_gdp_read_buffer): Made cancellable, actually returns the
whole buffer, and better diagnostics.
(gst_tcp_gdp_read_caps): Same.
* gst/sine/gstsinesrc.c (gst_sinesrc_wait): Add the base time.
Original commit message from CVS:
2005-09-26 Andy Wingo <wingo@pobox.com>
* gst/sine/gstsinesrc.h:
* gst/sine/gstsinesrc.c: Refactor, remove the table lookup code,
change the 'sync' property to 'is-live' and implement it halfway,
update for controller api change.
* gst/volume/gstvolume.c (volume_transform_ip): Update for
controller api change.
Original commit message from CVS:
* gst/audioresample/Makefile.am:
* gst/audioresample/debug.h:
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/resample.c: Convert to using gst debugging
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
(gst_play_bin_send_event):
Only seek on one sink, the first one that succeeds.
Original commit message from CVS:
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_clear),
(gst_vorbisenc_sink_event), (gst_vorbisenc_change_state):
Don't flush encoder state unless we have an initialised encoder.
Clear out encoder state on PAUSED_TO_READY.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init), (gst_basertppayload_init),
(gst_basertppayload_setcaps), (gst_basertppayload_chain),
(gst_basertppayload_set_options), (gst_basertppayload_set_outcaps),
(gst_basertppayload_is_filled), (gst_basertppayload_push),
(gst_basertppayload_set_property),
(gst_basertppayload_get_property),
(gst_basertppayload_change_state):
* gst-libs/gst/rtp/gstbasertppayload.h:
Added max-ptime to control amount of data in the rtp packets.
Original commit message from CVS:
2005-09-21 Andy Wingo <wingo@pobox.com>
* gst/playback/gstplaybasebin.c: Attempt to fix up buffer probe
thingies.
* gst/playback/gstdecodebin.c (gst_decode_bin_dispose): Dispose
can be called multiple times, dogs.
