Original commit message from CVS:
Patch by: Luis Menina <liberforce at freeside dot fr>
* gst-libs/gst/floatcast/floatcast.h:
* gst/typefind/gsttypefindfunctions.c:
Include glib.h instead of a specific GLib header. Including single
GLib headers is deprecated. Fixes bug #563904.
Original commit message from CVS:
2008-12-09 Julien Moutte <julien@fluendo.com>
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Support higher max audio rates for some formats (WAV, Vorbis, LPCM).
Original commit message from CVS:
Patch by: Olivier Crete <tester at tester ca>
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_remove):
* gst-libs/gst/rtp/gstrtcpbuffer.h:
Implement gst_rtcp_packet_remove(). Fixes#563174.
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
Add unit test for some RTCP functions.
Original commit message from CVS:
Patch by: 이문형 <iwings at gmail dot com>
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_push_buffer):
Don't forget to release the lock again if we bail out because some
pad is flushing or we've reached EOS, otherwise things will lock up
next time _push_buffer() is called (#562802).
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_get_offset), (gst_base_audio_src_create):
Avoid nasty int overflows after about 12 hours and 25 minutes when these
code paths are triggered.
A free beer to Håvard Graff for finding this!
Original commit message from CVS:
Patch by: 이문형 <iwings at gmail dot com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_connect):
A successful gst_poll_wait() doesn't always mean successful connect() on
Windows. We should check errors by calling gst_poll_fd_has_error().
See #561924.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_event),
(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
Really fix audiosink drain handling by keeping track of the running_time
of the last sample.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
Time is already in running_time. Remove base_time handling. Fixes
audiosinks not draining and thus chopping some audio in the end.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
Add one log message to check for audio_drained. Sync one log message
with the condition. Send EOS after draining audio in pull mode.
Original commit message from CVS:
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
(gst_rtp_buffer_set_packet_len), (gst_rtp_buffer_get_packet_len),
(gst_rtp_buffer_get_header_len), (gst_rtp_buffer_get_version),
(gst_rtp_buffer_set_version), (gst_rtp_buffer_get_padding),
(gst_rtp_buffer_set_padding), (gst_rtp_buffer_pad_to),
(gst_rtp_buffer_get_extension), (gst_rtp_buffer_set_extension),
(gst_rtp_buffer_get_extension_data),
(gst_rtp_buffer_set_extension_data), (gst_rtp_buffer_get_ssrc),
(gst_rtp_buffer_set_ssrc), (gst_rtp_buffer_get_csrc_count),
(gst_rtp_buffer_get_csrc), (gst_rtp_buffer_set_csrc),
(gst_rtp_buffer_get_marker), (gst_rtp_buffer_set_marker),
(gst_rtp_buffer_get_payload_type),
(gst_rtp_buffer_set_payload_type), (gst_rtp_buffer_get_seq),
(gst_rtp_buffer_set_seq), (gst_rtp_buffer_get_timestamp),
(gst_rtp_buffer_set_timestamp),
(gst_rtp_buffer_get_payload_subbuffer),
(gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload):
Avoid expensive type checks we already did as part of the
_validate() function that should be called first.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c: (create_segment_event),
(gst_base_rtp_depayload_push_full),
(gst_base_rtp_depayload_set_gst_timestamp):
Fix some cases where a newsegment event was not sent.
Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsrc.c:
* gst/h264parse/gsth264parse.c:
Wim, you're a bad boy. You don't want people to contact you or what?
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
(gst_base_audio_sink_callback):
Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting
for the latency to expire, fixes#559567.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Fix case where we don't have a range for the rates or channels as is the
case with truespeech.
Original commit message from CVS:
Patch by: Damien Lespiau <damien.lespiau gmail com>
* gst-libs/gst/rtsp/gstrtspconnection.c:
(gst_rtsp_connection_write):
Make the next call to poll not depend on previous calls to poll with or
without reading from the active descriptor. Fixes#544293.
Original commit message from CVS:
Patch by: Nick Haddad <nick at haddads dot net>
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Add support for other fourcc codes that are commonly used for
'uncompressed RGB', including 'RGB ', 'RAW ', and 0.
Fixes#558553.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/floatcast/floatcast.h:
Move float endianness conversion macros to core. Second part of
bug ##555196.
Original commit message from CVS:
* gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
Remove useless buffer size assignment. It already has this value.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_class_init), (gst_audioringbuffer_acquire),
(gst_audioringbuffer_activate), (gst_audioringbuffer_release),
(gst_audioringbuffer_stop):
Implement a separate activate functions to start monitoring the segments
or, in pull mode, pulling in data.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_init), (gst_base_audio_sink_dispose),
(gst_base_audio_sink_query_pad), (gst_base_audio_sink_query),
(gst_base_audio_sink_setcaps), (gst_base_audio_sink_callback),
(gst_base_audio_sink_activate_pull),
(gst_base_audio_sink_async_play),
(gst_base_audio_sink_change_state):
Implement pad and element convert query function.
Activate the ringbuffer.
Use the segment last_stop value as the offset to pull.
Use new basesink _do_preroll() method to preroll in the pulling thread.
Take appropriate locking in the pulling thread.
* gst-libs/gst/audio/gstringbuffer.h:
Update some docs.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_convert),
(gst_ring_buffer_activate), (gst_ring_buffer_is_active):
* gst-libs/gst/audio/gstringbuffer.h:
Add methods to more accuratly control the pulling thread of a
ringbuffer.
Add format conversion helper code to the ringbuffer.
API: GstRingBuffer:gst_ring_buffer_activate()
API: GstRingBuffer:gst_ring_buffer_is_active()
API: GstRingBuffer:gst_ring_buffer_convert()
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
(gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
(gst_audioringbuffer_stop):
Signal thread startup earlier so that we can immediatly go into pull
mode when we have to and block on preroll.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_prepare_read):
In pull mode we want the callback to prepull a buffer we can preroll on
even when we are not yet playing.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add mappping for the KMVC (Karl Morton's Video) Codec.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
(gst_base_rtp_depayload_set_gst_timestamp),
(gst_base_rtp_depayload_change_state):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Add some more G_LIKELY
Fail when the setcaps function was not called.
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_set_outcaps):
Propagate return value of setcaps.
Original commit message from CVS:
* gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
Don't drop the last byte of image tags if they're not an URI list.
Fixes bug #556066.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Fix debug statements (space between '%' and actual format).
Original commit message from CVS:
Patch by: Jan Gerber <j at oil21 dot org>
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add FFV1 fourcc to support playback of FFMPEG lossless video
in AVI. Fixes bug #555319.
Original commit message from CVS:
Based on Patch by: Tero Saarni <tero dot saarni at gmail dot com>
* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_parse_line),
(print_media), (gst_sdp_message_dump):
Fix parsing of the c= field containing multicast addresses.
Fixes#552199.
Add the connection info to the session or streams.
Fix parsing of the bandwidth.
Add debugging for the connections and bandwidths for a media.
Add debugging for the bandwidth of the session.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_change_state):
Configure the next seqnum and timestamp in the state change so that they
can be queried soon after.
Original commit message from CVS:
* gst-libs/gst/rtsp/gstrtspmessage.c:
(gst_rtsp_message_parse_request),
(gst_rtsp_message_parse_response):
Fix the g_return_val_if_fail() statements.
Original commit message from CVS:
* gst-libs/gst/tag/gsttagdemux.c:
Fail to activate if there's insufficient data in the file to be usable,
preventing an assertion fail later. Fixes#552960
Original commit message from CVS:
* gst-libs/gst/pbutils/descriptions.c:
* gst/typefind/gsttypefindfunctions.c:
Recognise Kate subtitle streams (#550582).
Original commit message from CVS:
* gst-libs/gst/audio/audio.h: (GST_AUDIO_FIELD_SIGNED):
Remove trailing comma from enum list, which causes problems
with -pendantic (#550729).
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_stop):
Disable a code path that is now called but causes a deadlock for some
reason and is unneeded.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Use a decent caps for TrueSpeech instead of a ffmpeg-specific one.
This will also be fixed for upcoming gst-ffmpeg release so that once
this release of -base is out, it will work with the latest gst-ffmpeg
release.
