Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
(gst_ring_buffer_read), (gst_ring_buffer_clear):
Don't assert on normal stuff.
* gst/playback/gstplaybin.c: (do_playbin_seek):
API fix.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
(gst_ring_buffer_read):
Cleanups.
Commit and read from ringbuffer in samples rather than bytes.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Respect segment rate and accum when scheduling samples.
Original commit message from CVS:
2005-10-11 Julien MOUTTE <julien@moutte.net>
* ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
(gst_ogg_mux_collected): Quick hack to fix build. We need to
handle
EOS correctly, that needs more work.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gst_play_bin_send_event_to_sink),
(do_playbin_seek), (gst_play_bin_send_event):
Override send_event differently, so that we can takes bits of
functionality from GstPipeline (special handling for seeks,
including pausing/resuming, and resetting stream time) and
still get
the appropriate behaviour of only forwarding event to a single
sink,
rather than all of them.
Unfortunately requires a lot of code duplication, but the
alternatives are equally ugly in the end.
Original commit message from CVS:
* check/elements/audioconvert.c: (setup_audioconvert),
(cleanup_audioconvert), (get_int_caps), (verify_convert),
(GST_START_TEST), (audioconvert_suite):
clean up tests a little, fix some leaks.
Original commit message from CVS:
* ext/alsa/gstalsasink.c:
Also allow unsigned int.
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
Small cleanup
Original commit message from CVS:
* check/pipelines/simple_launch_lines.c: (run_pipeline):
Small update, use API as stated in design docs.
* examples/seeking/seek.c: (make_avi_msmpeg4v3_mp3_pipeline),
(update_scale), (do_seek), (seek_cb), (set_update_scale),
(start_seek), (stop_seek), (play_cb), (pause_cb), (stop_cb),
(message_received), (main):
Updated seek example for GOption. Some usability improvements.
Original commit message from CVS:
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_unset_matrix),
(gst_channel_mix_setup_matrix), (gst_channel_mix_mix):
Alloc temp storage somewhere else where we can do it more
portable.
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.
