Original commit message from CVS:
* gst/wavenc/gstwavenc.c: (gst_wavenc_chain):
If we're not allowing width!=depth in wavenc we should also disable
the code that was added to support width!=depth.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream):
Don't calculate the default duration of a frame from the audio sampling
rate. This only works for raw audio if every frame contains a single
sample and results in broken buffer durations for other formats
if no specified default duration is given or the blocks have no
duration. Fixes bug #548831.
Original commit message from CVS:
* gst/matroska/matroska-demux.c:
(gst_matroska_demux_parse_blockgroup_or_simpleblock):
Allow zero sized blocks instead of returning GST_FLOW_OK. Such blocks
are used for text/plain subtitles as a gap-filler in some files.
Original commit message from CVS:
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_v4l2fourcc_to_structure),
(gst_v4l2_get_caps_info):
Add S910 and PWC formats with a low priority.
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_format_get_rank),
(gst_v4l2src_probe_caps_for_format):
Add more debugging.
Original commit message from CVS:
* gst/rtsp/gstrtspgoogle.c:
Things that can happen when your brain is in google mode trying to
deal with their google rtsp server extensions and trying to type your
google mail account.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtsp.c: (plugin_init):
* gst/rtsp/gstrtspgoogle.c: (gst_rtsp_google_before_send),
(gst_rtsp_google_after_send), (gst_rtsp_google_get_transports),
(_do_init), (gst_rtsp_google_base_init),
(gst_rtsp_google_class_init), (gst_rtsp_google_init),
(gst_rtsp_google_finalize), (gst_rtsp_google_change_state),
(gst_rtsp_google_extension_init):
* gst/rtsp/gstrtspgoogle.h:
Add google RTSP extension, it can only handle udp and responds with
unsupported if we do anything else. Fixes#546465.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_connection_send),
(gst_rtspsrc_connection_receive), (gst_rtspsrc_loop_send_cmd),
(gst_rtspsrc_create_transports_string),
(gst_rtspsrc_setup_streams), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_pause):
Make transport setup code a bit better using GString.
Add some more debug.
Check for closed connections before doing anything on them.
Original commit message from CVS:
* ext/pulse/pulsesrc.c: (gst_pulsesrc_class_init),
(gst_pulsesrc_create_stream), (gst_pulsesrc_negotiate),
(gst_pulsesrc_prepare):
* ext/pulse/pulseutil.c: (gst_pulse_gst_to_channel_map),
(gst_pulse_channel_map_to_gst):
* ext/pulse/pulseutil.h:
If downstream provides no channel layout and >2 channels should be
used use the default layout that pulseaudio chooses and also
add this layout to the caps. Fixes bug #547258.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_udp_sink):
Don't try to configure RTCP back to the server when the server did not
give us a valid port number.
Original commit message from CVS:
* gst/videobox/gstvideobox.c: (gst_video_box_set_property):
Use new basetransform method to renegotiate. Fixes#544956.
* tests/icles/Makefile.am:
* tests/icles/videobox-test.c: (make_pipeline), (main):
Add videobox renegotiation example.
Original commit message from CVS:
* ext/pulse/pulsesink.c: (gst_pulsesink_prepare):
* ext/pulse/pulsesrc.c: (gst_pulsesrc_prepare):
The bytes_per_sample and silence_sample fields of the GstRingBufferSpec
are already filled with the correct values by
gst_ring_buffer_parse_caps() so there's no need to set them again
with wrong values.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes_pull),
(gst_avi_demux_read_subindexes_push):
Some AVI 2.0 (ODML) files don't respect the 'specifications' completely
and instead of using the 'ix##' nomenclature, use '##ix'.
They're still valid though, this fixes the duration and indexes for
virtually all the ODML files I have.
Original commit message from CVS:
* ext/pulse/pulsesink.c: (gst_pulsesink_class_init),
(gst_pulsesink_init), (gst_pulsesink_finalize),
(gst_pulsesink_set_volume), (gst_pulsesink_get_volume),
(gst_pulsesink_set_property), (gst_pulsesink_get_property),
(gst_pulsesink_prepare), (gst_pulsesink_change_state):
* ext/pulse/pulsesink.h:
Add "device-name" property to pulsesink too and currently commented
out and not working support for a "volume" property.
Original commit message from CVS:
Patch by: Laszlo Pandy <laszlok2 at gmail dot com>
* ext/pulse/pulsesrc.c: (gst_pulsesrc_class_init),
(gst_pulsesrc_get_property):
Add "device-name" property, which provides a human readable string
for the audio device, to make it more consisten with other audio
sources. Fixes bug #547519.
Original commit message from CVS:
* ext/pulse/pulsemixer.c: (gst_pulsemixer_change_state):
* ext/pulse/pulsemixerctrl.c: (gst_pulsemixer_ctrl_subscribe_cb),
(gst_pulsemixer_ctrl_open), (gst_pulsemixer_ctrl_new),
(gst_pulsemixer_ctrl_free), (gst_pulsemixer_ctrl_timeout_event):
* ext/pulse/pulsemixerctrl.h:
* ext/pulse/pulseprobe.c: (gst_pulseprobe_open),
(gst_pulseprobe_enumerate), (gst_pulseprobe_new),
(gst_pulseprobe_free), (gst_pulseprobe_needs_probe),
(gst_pulseprobe_probe_property), (gst_pulseprobe_get_values):
* ext/pulse/pulseprobe.h:
* ext/pulse/pulsesink.c: (gst_pulsesink_init):
* ext/pulse/pulsesrc.c: (gst_pulsesrc_init), (gst_pulsesrc_delay),
(gst_pulsesrc_change_state):
Improve debugging a bit by including the parent object in pulsemixerctrl
and pulseprobe objects and using GST_WARNING_OBJECT instead of
GST_WARNING.
Use the parent GObject subclass instead of a random struct as GObject
parameter for G_OBJECT_WARN_INVALID_PROPERTY_ID. This fixes a crash
when probing for another property than "device".
Original commit message from CVS:
Patch by: Laszlo Pandy <laszlok2 at gmail dot com>
* ext/pulse/pulsemixer.c: (gst_pulsemixer_set_property):
Fix property probing after the device property is set by calling
set_server when the server property changes. Fixes bug #547518.
Original commit message from CVS:
Patch by: Laszlo Pandy <laszlok2 at gmail dot com>
* ext/pulse/pulsemixer.c: (gst_pulsemixer_set_property):
Fix property probing after the device property is set by calling
set_server when the server property changes. Fixes bug #547518.
Original commit message from CVS:
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_query):
Properly set the maximum latency value, in the same way it is done in
v4lsrc.
* sys/v4l2/v4l2src_calls.c:
Simplify fraction equality check, no need to use GValues for this.
Original commit message from CVS:
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_query):
Add warning messages stating exactly why the latency query failed.
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_set_capture):
In some cases, the negotiated framerate might be the default one which
is already set internally. But we still need to mark it down in fps_n
and fps_d so that the latency query can happen properly.
Original commit message from CVS:
* docs/plugins/inspect/plugin-1394.xml:
Whoops, forgot one doc file for people who can't/don't build the
raw1394 plugin.
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_message_new):
Fix compilation (also known as the classic 'fix code that someone
committed without compiling it first').
Original commit message from CVS:
* tests/examples/spectrum/demo-audiotest.c:
* tests/examples/spectrum/demo-osssrc.c:
Demo how to draw analyzer results synced to the clock.
Original commit message from CVS:
* gst/level/gstlevel.c:
Little renaming (l -> level).
* gst/spectrum/gstspectrum.c:
* gst/spectrum/gstspectrum.h:
Also send full timestamp/duration details here.
Original commit message from CVS:
* gst/level/gstlevel.c:
* gst/level/gstlevel.h:
Send same timestamp/duration details as videoanalysis. This gives
applications better chance to sync analysis results with playback.
Original commit message from CVS:
* gst/matroska/matroska-mux.c:
(gst_matroska_mux_handle_sink_event),
(flac_streamheader_to_codecdata):
We need to drop one additional buffer for FLAC as the fLaC
marker and STREAMINFO block are merged into one buffer in the caps.
Also don't pretend to support NEWSEGMENT events, otherwise we
will most probably write some invalid data.
Original commit message from CVS:
* gst/matroska/matroska-mux.c: (flac_streamheader_to_codecdata),
(gst_matroska_mux_audio_pad_setcaps):
Add support for muxing FLAC into Matroska containers.
Fixes bug #311586.
Original commit message from CVS:
* ext/flac/gstflacenc.c: (gst_flac_enc_check_discont):
Actually provide the variables required for the format string.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_reset),
(gst_matroska_demux_element_send_event),
(gst_matroska_demux_handle_seek_event), (gst_matroska_demux_loop):
* gst/matroska/matroska-demux.h:
Close the current segment if we're doing a non-flushing seek and send
the close-segment and the new segment of the seek from the streaming
thread.
Original commit message from CVS:
* ext/flac/gstflacenc.c: (gst_flac_enc_write_callback),
(gst_flac_enc_check_discont), (gst_flac_enc_chain),
(gst_flac_enc_change_state):
* ext/flac/gstflacenc.h:
Handle non-zero start timestamps correctly, mark header packets as
IN_CAPS and print a warning and suggest using audiorate if stream
discontinuities are detected. When FLAC supports flushing the encoder
somehow this should be done for discontinuities instead.
Remove some unused variables from the instance struct.
Original commit message from CVS:
* ext/flac/gstflacenc.c: (gst_flac_enc_seek_callback):
If seeking failed return the appropiate return value to FLAC.
Otherwise it thinks seeking was successfull and tries to rewrite
parts of the headers which then get appended to the output.
Original commit message from CVS:
Patch by: Frederic Crozat <fcrozat@mandriva.org>
* ext/esd/gstesd.c: (plugin_init):
* ext/flac/gstflac.c: (plugin_init):
* ext/shout2/gstshout2.c: (plugin_init):
* ext/wavpack/gstwavpack.c: (plugin_init):
* sys/oss/gstossaudio.c: (plugin_init):
* sys/v4l2/gstv4l2.c: (plugin_init):
Make sure gettext returns translations in UTF-8 encoding rather
than in the current locale encoding (#546822).
Original commit message from CVS:
* ext/flac/gstflacdec.c:
Add FIXME for 0.11 to simply output everything with width=32 as given
by FLAC and let audioconvert handle the conversions instead of doing
them in flacdec.
Original commit message from CVS:
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_probe_caps_for_format):
When outputting a pad template range for the size, include a framerate
range too, to avoid 'not a real subset of template caps' errors.
Original commit message from CVS:
Based on a patch by: Jonathan Matthew <notverysmart at gmail dot com>
* ext/flac/Makefile.am:
* ext/flac/gstflac.c: (plugin_init):
* ext/flac/gstflactag.c: (gst_flac_tag_setup_interfaces),
(gst_flac_tag_base_init), (gst_flac_tag_class_init),
(gst_flac_tag_dispose), (gst_flac_tag_init),
(gst_flac_tag_sink_setcaps), (gst_flac_tag_chain),
(gst_flac_tag_change_state):
* ext/flac/gstflactag.h:
Port flactag to 0.10, add documentation for it and clean it up a bit.
Fixes bug #413841.
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.hierarchy:
* docs/plugins/gst-plugins-good-plugins.interfaces:
* docs/plugins/gst-plugins-good-plugins.prerequisites:
* docs/plugins/inspect/plugin-flac.xml:
* ext/flac/gstflacdec.c: (gst_flac_dec_base_init):
* ext/flac/gstflacdec.h:
* ext/flac/gstflacenc.c: (gst_flac_enc_base_init):
* ext/flac/gstflacenc.h:
Add flactag and flacenc to the documentation and mark
the private parts of the flacdec instance structure as private.
Also use gst_element_class_set_details_simple() in flacdec and
flacenc.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c:
Use audio/x-qdm for caps. Collect some info - mplayer has a decoder
for it but ffmpeg does not.
